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[Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]


 
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mbgatherer at gmail.com
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PostPosted: Thu Nov 12, 2020 7:29 pm    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Hi all,


I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
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brians at iptel.co
Guest





PostPosted: Sat Nov 14, 2020 3:16 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
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david.villasmil.work a...
Guest





PostPosted: Fri Nov 20, 2020 6:13 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
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https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337
Back to top
mbgatherer at gmail.com
Guest





PostPosted: Fri Nov 20, 2020 6:18 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks
Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
Back to top
mbgatherer at gmail.com
Guest





PostPosted: Fri Nov 20, 2020 7:56 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

We normally use "proxy_media=true" and this is the situation that triggers the issue.IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
Back to top
brian at freeswitch.com
Guest





PostPosted: Fri Nov 20, 2020 11:53 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue.IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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_________________________________________________________________________

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Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
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https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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https://freeswitch.com

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FreeSWITCH-users mailing list
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--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]
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PostPosted: Sat Nov 21, 2020 2:46 pm    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Sure, will try...

Thanks
MAciej


pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com (brian@freeswitch.com)> napisał(a):

Quote:
set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue.IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]













_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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PostPosted: Tue Nov 24, 2020 2:39 pm    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

------ Start of attached email. Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host] ------
Try to add this line in the vars.xml

<action application="set" data="media_mix_inbound_outbound_codecs=true"/>

Regards,
Sonny

Quote:
On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer@gmail.com> wrote:


Sure, will try...

Thanks
MAciej

pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com> napisał(a):
Quote:
set them to "autonat:x.x.x.x" then set local-network-acl


Quote:
On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com> wrote:
We normally use "proxy_media=true" and this is the situation that triggers the issue.
IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>

Thanks
Maciej.

pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com> napisał(a):
Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?

Quote:
On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com> wrote:
Hello,

Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.

Thanks
Maciej



sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co> napisał(a):
Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20;user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

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_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
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Official FreeSWITCH Sites
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Regards,

David Villasmil
email: david.villasmil.work@gmail.com
phone: +34669448337
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

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Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

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FreeSWITCH-users@lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com


------ End of attached email ------
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
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https://freeswitch.org/confluence
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mbgatherer at gmail.com
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PostPosted: Sat Nov 28, 2020 9:23 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Hi

Will check that one and update you


Thanks
Maciej.


wt., 24 lis 2020 o 19:42 Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)> napisał(a):

Quote:



---------- Forwarded message ----------
From: Sonny Lafuente <lafuente_sonny@yahoo.com (lafuente_sonny@yahoo.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Sun, 22 Nov 2020 21:21:02 +0800
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
Try to add this line in the vars.xml 

Quote:
<action application="set" data="media_mix_inbound_outbound_codecs=true"/>

Regards,Sonny


Quote:
On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Sure, will try...

Thanks
MAciej


pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com (brian@freeswitch.com)> napisał(a):

Quote:
set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue.IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
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Build your next product on our scalable cloud platform.

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Join our online community to chat in real time https://signalwire.community

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--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]













_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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_________________________________________________________________________

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Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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---------- Forwarded message ----------
From: Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Tue, 24 Nov 2020 10:42:02 -0800 (PST)
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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mbgatherer at gmail.com
Guest





PostPosted: Sun Nov 29, 2020 3:10 pm    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Hi,

Unfortunately neither option helped.
- tried to set autonat:MY_PUBLIC_IP and local-network-acl
- added media_mix_inbound_outbound_codecs=true into dialplan/vars


I saw that putting <action application="set" data="bypass_media=true"/> helps to workaround this issue, but this is not exactly what I am looking for.
Frankly I do not think changing anything relating to NAT would help as my server as well as both end servers (traffic originator and terminator) use public IPs (logs i attached have got modified/not-real IPs).


Do you have any other clues, how to track the problem down in order to solve it?


Thanks
Maciej.






sob., 28 lis 2020 o 14:56 Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> napisał(a):

Quote:
Hi

Will check that one and update you


Thanks
Maciej.


wt., 24 lis 2020 o 19:42 Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)> napisał(a):

Quote:



---------- Forwarded message ----------
From: Sonny Lafuente <lafuente_sonny@yahoo.com (lafuente_sonny@yahoo.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Sun, 22 Nov 2020 21:21:02 +0800
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
Try to add this line in the vars.xml 

Quote:
<action application="set" data="media_mix_inbound_outbound_codecs=true"/>

Regards,Sonny


Quote:
On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Sure, will try...

Thanks
MAciej


pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com (brian@freeswitch.com)> napisał(a):

Quote:
set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue.IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
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_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
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https://cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]













_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com





---------- Forwarded message ----------
From: Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Tue, 24 Nov 2020 10:42:02 -0800 (PST)
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
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shaun at sysconfig.cloud
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PostPosted: Mon Nov 30, 2020 6:06 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Have you tried restarting FreeSWITCH?


It's possibly unrelated to your issue but we've recently noticed an increase in 'INCOMPATIBLE_DESTINATION' errors in our environment. We use 'media_mix_inbound_outbound_codecs=true' set as a global variable in '/conf/vars.xml' which works but it seems that over long periods of time FreeSWITCH begins to ignore this, restarting FreeSWITCH resolves the issue for us at least temporarily.


Thanks,
Shaun
From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Maciej Bylica <mbgatherer@gmail.com>
Sent: 29 November 2020 20:56
To: Sonny Lafuente <lafuente_sonny@yahoo.com>; FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]

Hi,

Unfortunately neither option helped.
- tried to set autonat:MY_PUBLIC_IP and local-network-acl
- added media_mix_inbound_outbound_codecs=true into dialplan/vars


I saw that putting <action application="set" data="bypass_media=true"/> helps to workaround this issue, but this is not exactly what I am looking for.
Frankly I do not think changing anything relating to NAT would help as my server as well as both end servers (traffic originator and terminator) use public IPs (logs i attached have got modified/not-real IPs).


Do you have any other clues, how to track the problem down in order to solve it?


Thanks
Maciej.






sob., 28 lis 2020 o 14:56 Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> napisał(a):

Quote:
Hi

Will check that one and update you


Thanks
Maciej.


wt., 24 lis 2020 o 19:42 Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)> napisał(a):

Quote:



---------- Forwarded message ----------
From: Sonny Lafuente <lafuente_sonny@yahoo.com (lafuente_sonny@yahoo.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc:
Bcc:
Date: Sun, 22 Nov 2020 21:21:02 +0800
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
Try to add this line in the vars.xml

Quote:
<action application="set" data="media_mix_inbound_outbound_codecs=true"/>

Regards, Sonny


Quote:
On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

 Sure, will try...

Thanks
MAciej


pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com (brian@freeswitch.com)> napisał(a):

Quote:
set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue. IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
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--
Regards,


David Villasmil email: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
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FreeSWITCH-users mailing list
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--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]













_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com





---------- Forwarded message ----------
From: Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc:
Bcc:
Date: Tue, 24 Nov 2020 10:42:02 -0800 (PST)
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
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mbgatherer at gmail.com
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PostPosted: Mon Nov 30, 2020 7:48 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

Hi Shaun,

Yes, FreeSWITCH was restarted every time I changed something in the configuration.
I was flooded with "AUDIO RTP REPORTS ERROR: [Missing local host]" and the following "INCOMPATIBLE_DESTINATION" logs the same way.


Thanks
Maciej


pon., 30 lis 2020 o 12:44 Shaun Stokes <shaun@sysconfig.cloud> napisał(a):

Quote:
Have you tried restarting FreeSWITCH?


It's possibly unrelated to your issue but we've recently noticed an increase in 'INCOMPATIBLE_DESTINATION' errors in our environment. We use 'media_mix_inbound_outbound_codecs=true' set as a global variable in '/conf/vars.xml' which works but it seems that over long periods of time FreeSWITCH begins to ignore this, restarting FreeSWITCH resolves the issue for us at least temporarily.


Thanks,
Shaun
From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)> on behalf of Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)>
Sent: 29 November 2020 20:56
To: Sonny Lafuente <lafuente_sonny@yahoo.com (lafuente_sonny@yahoo.com)>; FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]  

Hi,

Unfortunately neither option helped.
- tried to set autonat:MY_PUBLIC_IP and local-network-acl
- added media_mix_inbound_outbound_codecs=true into dialplan/vars


I saw that putting <action application="set" data="bypass_media=true"/> helps to workaround this issue, but this is not exactly what I am looking for.
Frankly I do not think changing anything relating to NAT would help as my server as well as both end servers (traffic originator and terminator) use public IPs (logs i attached have got modified/not-real IPs).


Do you have any other clues, how to track the problem down in order to solve it?


Thanks
Maciej.






sob., 28 lis 2020 o 14:56 Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> napisał(a):

Quote:
Hi

Will check that one and update you


Thanks
Maciej.


wt., 24 lis 2020 o 19:42 Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)> napisał(a):

Quote:



---------- Forwarded message ----------
From: Sonny Lafuente <lafuente_sonny@yahoo.com (lafuente_sonny@yahoo.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Sun, 22 Nov 2020 21:21:02 +0800
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
Try to add this line in the vars.xml 

Quote:
<action application="set" data="media_mix_inbound_outbound_codecs=true"/>

Regards, Sonny


Quote:
On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

 Sure, will try...

Thanks
MAciej


pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com (brian@freeswitch.com)> napisał(a):

Quote:
set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue. IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

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_________________________________________________________________________

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Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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--
Regards,


David Villasmil email: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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--



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Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]













_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
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Official FreeSWITCH Sites
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https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
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---------- Forwarded message ----------
From: Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Tue, 24 Nov 2020 10:42:02 -0800 (PST)
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com





_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
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Back to top
brian at freeswitch.com
Guest





PostPosted: Mon Nov 30, 2020 10:39 am    Post subject: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO R Reply with quote

bypass isn't a fix, understanding the signaling and network path are key here, What did you set your local-network-acl to?  And are you sure the settings were applied after setting them?

/b




On Sun, Nov 29, 2020 at 2:21 PM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hi,

Unfortunately neither option helped.
- tried to set autonat:MY_PUBLIC_IP and local-network-acl
- added media_mix_inbound_outbound_codecs=true into dialplan/vars


I saw that putting <action application="set" data="bypass_media=true"/> helps to workaround this issue, but this is not exactly what I am looking for.
Frankly I do not think changing anything relating to NAT would help as my server as well as both end servers (traffic originator and terminator) use public IPs (logs i attached have got modified/not-real IPs).


Do you have any other clues, how to track the problem down in order to solve it?


Thanks
Maciej.






sob., 28 lis 2020 o 14:56 Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> napisał(a):

Quote:
Hi

Will check that one and update you


Thanks
Maciej.


wt., 24 lis 2020 o 19:42 Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)> napisał(a):

Quote:



---------- Forwarded message ----------
From: Sonny Lafuente <lafuente_sonny@yahoo.com (lafuente_sonny@yahoo.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Sun, 22 Nov 2020 21:21:02 +0800
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
Try to add this line in the vars.xml 

Quote:
<action application="set" data="media_mix_inbound_outbound_codecs=true"/>

Regards,Sonny


Quote:
On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Sure, will try...

Thanks
MAciej


pt., 20 lis 2020 o 17:47 Brian West <brian@freeswitch.com (brian@freeswitch.com)> napisał(a):

Quote:
set them to "autonat:x.x.x.x" then set local-network-acl



On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
We normally use "proxy_media=true" and this is the situation that triggers the issue.IPs provided are modified into 10.20.30.xx form, but we use public IPs, so no NAT is involved, hence both parameters have default values like:

    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>


Thanks
Maciej.


pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)> napisał(a):

Quote:
In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?


On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:

Quote:
Hello,


Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.


Thanks


Maciej









sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co (brians@iptel.co)> napisał(a):

Quote:
I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com (mbgatherer@gmail.com)> wrote:
Quote:
Hi all,
I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.
I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.

Some of the important log snippets:
- the call is CANCELed once FS receives 183 and gets first early media packets

8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20)] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 (4916222112233@10.20.30.20) Media Establishment Failed.
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)
8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.

- incoming (Originator -> FS) SIP INVITE has following SDP
v=0
o=- 1604689110 1604689110 IN IP4 10.20.30.6
s=-
c=IN IP4 10.20.30.6
t=0 0
m=audio 26502 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

- outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
v=0
o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
s=FreeSWITCH
c=IN IP4 10.20.30.20
t=0 0
m=audio 31034 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

- 183 that is received (Terminator -> FS)
v=0
o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
s=JOANO-SIP
c=IN IP4 79.133.196.167
t=0 0
m=audio 18200 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

As a result FS tears down the call attempt by sending (FS -> Originator)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
Max-Forwards: 66
From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
To: <sip:3144917111223344@10.20.30.20 ([email]sip%3A3144917111223344@10.20.30.20[/email]);user=phone>;tag=DcUe4343Uvj8Q
Call-ID: 684d5998-d859dd12-ca30a685-551b1244
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

and CANCEL (FS -> Terminator)
CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
Max-Forwards: 70
From: <sip:4916222112233@10.20.30.20 ([email]sip%3A4916222112233@10.20.30.20[/email])>;tag=g76r9mQeKQN0a
To: <sip:3144917111223344@10.20.30.30:5060>
Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
CSeq: 27790763 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

Could somebody help me to address the issue I am struggling with ?

Detailed debugging logs and sip signalization might be found here:
https://pastebin.com/G0sjGiw8 - FS logs
https://pastebin.com/2Zt6r2mF - SIP logs

Many thanks in advance,
Maciej
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--
Regards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]













_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com





---------- Forwarded message ----------
From: Sonny Lafuente via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Tue, 24 Nov 2020 10:42:02 -0800 (PST)
Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]
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