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[Freeswitch-users] Passthru mode


 
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fvillarroel at yahoo.com
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PostPosted: Mon Jun 01, 2009 11:55 am    Post subject: [Freeswitch-users] Passthru mode Reply with quote

Dear all.

I have problem with g729 passthru mode.

I received traffic from a Asterisk on my FS and forward to other Asterisk, when i use codec ulaw this works very well.

But when i try use G729 i received the following messages and SIP Trace:

2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/admin/42452904@190.208.xx.yy [f65514e0-4ec7-11de-9b78-150e2985561f]
2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/42452904@190.208.xx.yy entering state [received][100]
2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote SDP:
v=0
o=root 25643 25643 IN IP4 190.208.xx.yy
s=session
c=IN IP4 190.208.xx.yy
t=0 0
m=audio 10236 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20]
2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20]
2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup sofia/admin/42452904@190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION]
2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/42452904@190.208.xx.yy [KILL]
2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904@190.208.xx.yy [BREAK]
send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83
From: "102" <sip:102@200.111.XXX.XX>;tag=bmke36jc1v
To: "102" <sip:102@200.111.XXX.XX>;tag=rZp0XXrK9NHFD
Call-ID: 3c26700b249f-sryanqz0td8u@snom360-00041323143F
CSeq: 27056 REGISTER
Contact: <sip:102@192.168.1.124:2051;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:72e96588-ebe1-476d-8024-75656b4e007d>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30
Date: Mon, 01 Jun 2009 16:19:20 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Content-Length: 0

------------------------------------------------------------------------
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) Running State Change CS_HANGUP
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State HANGUP
2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/42452904@190.208.xx.yy hanging up, cause: INCOMPATIBLE_DESTINATION
2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 488
send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208:
------------------------------------------------------------------------
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060
From: "452904" <sip:42452904@190.208.xx.yy>;tag=as4e2616ae
To: <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r
Call-ID: 117330d21f3828470f39a95f538be036@190.208.xx.yy
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0

------------------------------------------------------------------------
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/42452904@190.208.xx.yy Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State HANGUP going to sleep
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State Change CS_HANGUP -> CS_REPORTING
2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904@190.208.xx.yy [BREAK]
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) Running State Change CS_REPORTING
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904@190.208.xx.yy) State REPORTING
recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338:
------------------------------------------------------------------------
ACK sip:56968482060@200.111.XXX.XX SIP/2.0
Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport
From: "452904" <sip:42452904@190.208.xx.yy>;tag=as4e2616ae
To: <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r
Contact: <sip:42452904@190.208.xx.yy>
Call-ID: 117330d21f3828470f39a95f538be036@190.208.xx.yy
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

------------------------------------------------------------------------
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/42452904@190.208.xx.yy Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904@190.208.xx.yy) State REPORTING going to sleep
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State Change CS_REPORTING -> CS_DESTROY
2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/admin/42452904@190.208.xx.yy) Locked, Waiting on external entities
2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/admin/42452904@190.208.xx.yy) Ended
2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/42452904@190.208.xx.yy [CS_DESTROY]
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904@190.208.xx.yy) State DESTROY
2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/42452904@190.208.xx.yy SOFIA DESTROY
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/42452904@190.208.xx.yy Standard DESTROY
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904@190.208.xx.yy) State DESTROY going to sleep

My vars.xml :

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/>
<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>


I hope your comments for know where is the config problem

Fernando.





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PostPosted: Mon Jun 01, 2009 12:17 pm    Post subject: [Freeswitch-users] Passthru mode Reply with quote

What does your dialplan look like? Just curious where/how you set proxy-media mode.
-MC

On Mon, Jun 1, 2009 at 9:54 AM, FERNANDO VILLARROEL <fvillarroel@yahoo.com (fvillarroel@yahoo.com)> wrote:
Quote:

Dear all.

I have problem with g729 passthru mode.

I received traffic from a Asterisk on my FS and forward to other Asterisk, when i use codec ulaw this works very well.

But when i try use G729 i received the following messages and SIP Trace:

2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/admin/42452904@190.208.xx.yy [f65514e0-4ec7-11de-9b78-150e2985561f]
2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/42452904@190.208.xx.yy entering state [received][100]
2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote SDP:
v=0
o=root 25643 25643 IN IP4 190.208.xx.yy
s=session
c=IN IP4 190.208.xx.yy
t=0 0
m=audio 10236 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20]
2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20]
2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup sofia/admin/42452904@190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION]
2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/42452904@190.208.xx.yy [KILL]
2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904@190.208.xx.yy [BREAK]
send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633:
  ------------------------------------------------------------------------
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83
  From: "102" <sip:102@200.111.XXX.XX>;tag=bmke36jc1v
  To: "102" <sip:102@200.111.XXX.XX>;tag=rZp0XXrK9NHFD
  Call-ID: 3c26700b249f-sryanqz0td8u@snom360-00041323143F
  CSeq: 27056 REGISTER
  Contact: <sip:102@192.168.1.124:2051;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:72e96588-ebe1-476d-8024-75656b4e007d>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30
  Date: Mon, 01 Jun 2009 16:19:20 GMT
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  Supported: timer, precondition, path, replaces
  Content-Length: 0

  ------------------------------------------------------------------------
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) Running State Change CS_HANGUP
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State HANGUP
2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/42452904@190.208.xx.yy hanging up, cause: INCOMPATIBLE_DESTINATION
2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 488
send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208:
  ------------------------------------------------------------------------
  SIP/2.0 488 Not Acceptable Here
  Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060
  From: "452904" <sip:42452904@190.208.xx.yy>;tag=as4e2616ae
  To: <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r
  Call-ID: 117330d21f3828470f39a95f538be036@190.208.xx.yy
  CSeq: 102 INVITE
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431
  Accept: application/sdp
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  Supported: timer, precondition, path, replaces
  Allow-Events: talk, refer
  Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
  Content-Length: 0

  ------------------------------------------------------------------------
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/42452904@190.208.xx.yy Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State HANGUP going to sleep
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State Change CS_HANGUP -> CS_REPORTING
2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904@190.208.xx.yy [BREAK]
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) Running State Change CS_REPORTING
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904@190.208.xx.yy) State REPORTING
recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338:
  ------------------------------------------------------------------------
  ACK sip:56968482060@200.111.XXX.XX SIP/2.0
  Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport
  From: "452904" <sip:42452904@190.208.xx.yy>;tag=as4e2616ae
  To: <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r
  Contact: <sip:42452904@190.208.xx.yy>
  Call-ID: 117330d21f3828470f39a95f538be036@190.208.xx.yy
  CSeq: 102 ACK
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Content-Length: 0

  ------------------------------------------------------------------------
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/42452904@190.208.xx.yy Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904@190.208.xx.yy) State REPORTING going to sleep
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/42452904@190.208.xx.yy) State Change CS_REPORTING -> CS_DESTROY
2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/admin/42452904@190.208.xx.yy) Locked, Waiting on external entities
2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/admin/42452904@190.208.xx.yy) Ended
2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/42452904@190.208.xx.yy [CS_DESTROY]
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904@190.208.xx.yy) State DESTROY
2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/42452904@190.208.xx.yy SOFIA DESTROY
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/42452904@190.208.xx.yy Standard DESTROY
2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904@190.208.xx.yy) State DESTROY going to sleep

My vars.xml :

 <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/>
 <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/>
 <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
 <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>


I hope your comments for know where is the config problem

Fernando.





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fvillarroel at yahoo.com
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PostPosted: Mon Jun 01, 2009 5:23 pm    Post subject: [Freeswitch-users] Passthru mode Reply with quote

Hello the dial plan:

<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

This i setup from Wikipbx.



--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org> wrote:

Quote:
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 2:15 PM
What does your dialplan look like? Just
curious where/how you set proxy-media mode.
-MC

On Mon, Jun 1, 2009 at 9:54 AM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:



Dear all.



I have problem with g729 passthru mode.



I received traffic from a Asterisk on my FS and forward to
other Asterisk, when i use codec ulaw this works very well.



But when i try use G729 i received the following messages
and SIP Trace:



2009-06-01 12:19:20 [NOTICE] switch_channel.c:602
switch_channel_set_name() New Channel
sofia/admin/42452904@190.208.xx.yy
[f65514e0-4ec7-11de-9b78-150e2985561f]

2009-06-01 12:19:20 [DEBUG] sofia.c:3037
sofia_handle_sip_i_state() Channel
sofia/admin/42452904@190.208.xx.yy entering state
[received][100]

2009-06-01 12:19:20 [DEBUG] sofia.c:3044
sofia_handle_sip_i_state() Remote SDP:

v=0

o=root 25643 25643 IN IP4 190.208.xx.yy

s=session

c=IN IP4 190.208.xx.yy

t=0 0

m=audio 10236 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -



2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955
sofia_glue_negotiate_sdp() Audio Codec Compare
[G729:18:8000:0]/[PCMU:0:8000:20]

2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915
sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101

2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955
sofia_glue_negotiate_sdp() Audio Codec Compare
[telephone-event:101:8000:0]/[PCMU:0:8000:20]

2009-06-01 12:19:20 [NOTICE] sofia.c:3246
sofia_handle_sip_i_state() Hangup
sofia/admin/42452904@190.208.xx.yy [CS_NEW]
[INCOMPATIBLE_DESTINATION]

2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660
switch_channel_perform_hangup() Send signal
sofia/admin/42452904@190.208.xx.yy [KILL]

2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933
switch_core_session_signal_state_change() Send signal
sofia/admin/42452904@190.208.xx.yy [BREAK]

send 886 bytes to udp/[190.47.91.83]:60245 at
16:19:20.596633:

 
------------------------------------------------------------------------

  SIP/2.0 200 OK

  Via: SIP/2.0/UDP
192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83

  From: "102"
<sip:102@200.111.XXX.XX>;tag=bmke36jc1v

  To: "102"
<sip:102@200.111.XXX.XX>;tag=rZp0XXrK9NHFD

  Call-ID:
3c26700b249f-sryanqz0td8u@snom360-00041323143F

  CSeq: 27056 REGISTER

  Contact:
<sip:102@192.168.1.124:2051;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:72e96588-ebe1-476d-8024-75656b4e007d>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30


  Date: Mon, 01 Jun 2009 16:19:20 GMT

  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431

  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO

  Supported: timer, precondition, path, replaces

  Content-Length: 0



 
------------------------------------------------------------------------

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run()
(sofia/admin/42452904@190.208.xx.yy) Running State Change
CS_HANGUP

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433
switch_core_session_run()
(sofia/admin/42452904@190.208.xx.yy) State HANGUP

2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323
sofia_on_hangup() Channel sofia/admin/42452904@190.208.xx.yy
hanging up, cause: INCOMPATIBLE_DESTINATION

2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399
sofia_on_hangup() Responding to INVITE with: 488

send 634 bytes to udp/[190.208.xx.yy]:5060 at
16:19:20.603208:

 
------------------------------------------------------------------------

  SIP/2.0 488 Not Acceptable Here

  Via: SIP/2.0/UDP
190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060

  From: "452904"
<sip:42452904@190.208.xx.yy>;tag=as4e2616ae

  To:
<sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r

  Call-ID:
117330d21f3828470f39a95f538be036@190.208.xx.yy

  CSeq: 102 INVITE

  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431

  Accept: application/sdp

  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO

  Supported: timer, precondition, path, replaces

  Allow-Events: talk, refer

  Reason:
Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"

  Content-Length: 0



 
------------------------------------------------------------------------

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/admin/42452904@190.208.xx.yy Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433
switch_core_session_run()
(sofia/admin/42452904@190.208.xx.yy) State HANGUP going to
sleep

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475
switch_core_session_run()
(sofia/admin/42452904@190.208.xx.yy) State Change CS_HANGUP
-> CS_REPORTING

2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933
switch_core_session_signal_state_change() Send signal
sofia/admin/42452904@190.208.xx.yy [BREAK]

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run()
(sofia/admin/42452904@190.208.xx.yy) Running State Change
CS_REPORTING

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607
switch_core_session_reporting_state()
(sofia/admin/42452904@190.208.xx.yy) State REPORTING

recv 408 bytes from udp/[190.208.xx.yy]:5060 at
16:19:20.620338:

 
------------------------------------------------------------------------

  ACK sip:56968482060@200.111.XXX.XX SIP/2.0

  Via: SIP/2.0/UDP
190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport

  From: "452904"
<sip:42452904@190.208.xx.yy>;tag=as4e2616ae

  To:
<sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r

  Contact: <sip:42452904@190.208.xx.yy>

  Call-ID:
117330d21f3828470f39a95f538be036@190.208.xx.yy

  CSeq: 102 ACK

  User-Agent: Asterisk PBX

  Max-Forwards: 70

  Content-Length: 0



 
------------------------------------------------------------------------

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53
switch_core_standard_on_reporting()
sofia/admin/42452904@190.208.xx.yy Standard REPORTING,
cause: INCOMPATIBLE_DESTINATION

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607
switch_core_session_reporting_state()
(sofia/admin/42452904@190.208.xx.yy) State REPORTING going
to sleep

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410
switch_core_session_run()
(sofia/admin/42452904@190.208.xx.yy) State Change
CS_REPORTING -> CS_DESTROY

2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067
switch_core_session_thread() Session 7
(sofia/admin/42452904@190.208.xx.yy) Locked, Waiting on
external entities

2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085
switch_core_session_thread() Session 7
(sofia/admin/42452904@190.208.xx.yy) Ended

2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087
switch_core_session_thread() Close Channel
sofia/admin/42452904@190.208.xx.yy [CS_DESTROY]

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559
switch_core_session_destroy_state()
(sofia/admin/42452904@190.208.xx.yy) State DESTROY

2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240
sofia_on_destroy() sofia/admin/42452904@190.208.xx.yy SOFIA
DESTROY

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60
switch_core_standard_on_destroy()
sofia/admin/42452904@190.208.xx.yy Standard DESTROY

2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559
switch_core_session_destroy_state()
(sofia/admin/42452904@190.208.xx.yy) State DESTROY going to
sleep



My vars.xml :



 <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/>

 <X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/>

 <X-PRE-PROCESS cmd="set"
data="xmpp_client_profile=xmppc"/>

 <X-PRE-PROCESS cmd="set"
data="xmpp_server_profile=xmpps"/>





I hope your comments for know where is the config problem



Fernando.











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PostPosted: Mon Jun 01, 2009 5:42 pm    Post subject: [Freeswitch-users] Passthru mode Reply with quote

On Mon, Jun 1, 2009 at 3:20 PM, FERNANDO VILLARROEL <fvillarroel@yahoo.com (fvillarroel@yahoo.com)> wrote:
Quote:

Hello the dial plan:

<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

This i setup from Wikipbx.


What about this in the dialplan?
Quote:
<action application="set" data="proxy_media=true"/>
Or alternatively this in the SIP profile?
Quote:
<param name="inbound-proxy-media" value="true"/>

I just want to make sure you're actually telling FS to use proxy media. If I may make a suggestion: use pastebin.freeswitch.org and pastebin the entire extension in the dialplan as well as a complete debug log of the call from the FS CLI. Please see this page for some handy tips on gathering information for troubleshooting:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC
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PostPosted: Mon Jun 01, 2009 6:11 pm    Post subject: [Freeswitch-users] Passthru mode Reply with quote

Hello i was try with:

<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

This is the log on FS_CLI:

http://pastebin.freeswitch.org/9204

Fernando

--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org> wrote:

Quote:
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 7:41 PM


On Mon, Jun 1, 2009 at 3:20 PM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:



Hello the dial plan:



<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>



This i setup from Wikipbx.
What about this in the dialplan?
<action application="set"
data="proxy_media=true"/>
Or alternatively this in the SIP profile?

<param name="inbound-proxy-media"
value="true"/>

I just want to make sure you're actually telling FS to
use proxy media. If I may make a suggestion: use pastebin.freeswitch.org
and pastebin the entire extension in the dialplan as well as
a complete debug log of the call from the FS CLI. Please see
this page for some handy tips on gathering information for
troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC



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PostPosted: Mon Jun 01, 2009 6:11 pm    Post subject: [Freeswitch-users] Passthru mode Reply with quote

Hello i was try with:

<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

This is the log on FS_CLI:

http://pastebin.freeswitch.org/9204

Fernando

--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org> wrote:

Quote:
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 7:41 PM


On Mon, Jun 1, 2009 at 3:20 PM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:



Hello the dial plan:



<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>



This i setup from Wikipbx.
What about this in the dialplan?
<action application="set"
data="proxy_media=true"/>
Or alternatively this in the SIP profile?

<param name="inbound-proxy-media"
value="true"/>

I just want to make sure you're actually telling FS to
use proxy media. If I may make a suggestion: use pastebin.freeswitch.org
and pastebin the entire extension in the dialplan as well as
a complete debug log of the call from the FS CLI. Please see
this page for some handy tips on gathering information for
troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC



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PostPosted: Tue Jun 02, 2009 3:59 pm    Post subject: [Freeswitch-users] Passthru mode Reply with quote

Dear,

I can't solve my problem, i was try with:

<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

and:

<param name="disable-transcoding" value="true"/> in freeswitch.xml

But receive the same log:

http://pastebin.freeswitch.org/9204

Anyone help me.

Fernando

--- On Mon, 6/1/09, FERNANDO VILLARROEL <fvillarroel@yahoo.com> wrote:

Quote:
From: FERNANDO VILLARROEL <fvillarroel@yahoo.com>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 8:10 PM

Hello i was try with:

<action application="set" data="bypass_media=true"/>
<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>

This is the log on FS_CLI:

http://pastebin.freeswitch.org/9204

Fernando

--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org>
wrote:

Quote:
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 7:41 PM


On Mon, Jun 1, 2009 at 3:20 PM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:



Hello the dial plan:



<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>



This i setup from Wikipbx.
What about this in the dialplan?
<action application="set"
data="proxy_media=true"/>
Or alternatively this in the SIP profile?

<param name="inbound-proxy-media"
value="true"/>

I just want to make sure you're actually telling FS
to
Quote:
use proxy media. If I may make a suggestion: use
pastebin.freeswitch.org
Quote:
and pastebin the entire extension in the dialplan as
well as
Quote:
a complete debug log of the call from the FS CLI.
Please see
Quote:
this page for some handy tips on gathering information
for
Quote:
troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC



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PostPosted: Wed Jun 03, 2009 2:16 am    Post subject: [Freeswitch-users] Passthru mode Reply with quote

Fernando,

Try setting 'inbound-late-negotiation' in your SIP Profile. This will
allow the call to hit the dialplan where you can set proxy_media.
This also assumes you have bypass_media set to false in your dialplan.

Alternatively I beleive you can set "inbound-proxy-media" in the SIP
Profile and this will do the same thing.

Regards,
Jim

On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL
<fvillarroel@yahoo.com> wrote:
Quote:

Dear,

I can't solve my problem, i was try with:

<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

and:

<param name="disable-transcoding" value="true"/> in freeswitch.xml

But receive the same log:

http://pastebin.freeswitch.org/9204

Anyone help me.

Fernando

--- On Mon, 6/1/09, FERNANDO VILLARROEL <fvillarroel@yahoo.com> wrote:

Quote:
From: FERNANDO VILLARROEL <fvillarroel@yahoo.com>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 8:10 PM

Hello i was try with:

<action application="set" data="bypass_media=true"/>
<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>

This is the log on FS_CLI:

http://pastebin.freeswitch.org/9204

Fernando

--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org>
wrote:

Quote:
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 7:41 PM


On Mon, Jun 1, 2009 at 3:20 PM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:



Hello the dial plan:



<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>



This i setup from Wikipbx.
What about this in the dialplan?
<action application="set"
data="proxy_media=true"/>
Or alternatively this in the SIP profile?

<param name="inbound-proxy-media"
value="true"/>

I just want to make sure you're actually telling FS
to
Quote:
use proxy media. If I may make a suggestion: use
pastebin.freeswitch.org
Quote:
and pastebin the entire extension in the dialplan as
well as
Quote:
a complete debug log of the call from the FS CLI.
Please see
Quote:
this page for some handy tips on gathering information
for
Quote:
troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC



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PostPosted: Wed Jun 03, 2009 2:49 am    Post subject: [Freeswitch-users] Passthru mode Reply with quote

On 3-Jun-09, at 2:32 AM, Jim Burke wrote:

Quote:
Fernando,

Try setting 'inbound-late-negotiation' in your SIP Profile. This will
allow the call to hit the dialplan where you can set proxy_media.
This also assumes you have bypass_media set to false in your dialplan.

Alternatively I beleive you can set "inbound-proxy-media" in the SIP
Profile and this will do the same thing.

But you still need late negotiation for that to work, so in both cases
you need to fix that Very Happy

Math

Quote:


Regards,
Jim

On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL
<fvillarroel@yahoo.com> wrote:
Quote:

Dear,

I can't solve my problem, i was try with:

<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

and:

<param name="disable-transcoding" value="true"/> in freeswitch.xml

But receive the same log:

http://pastebin.freeswitch.org/9204

Anyone help me.

Fernando

--- On Mon, 6/1/09, FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:

Quote:
From: FERNANDO VILLARROEL <fvillarroel@yahoo.com>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 8:10 PM

Hello i was try with:

<action application="set" data="bypass_media=true"/>
<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>

This is the log on FS_CLI:

http://pastebin.freeswitch.org/9204

Fernando

--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org>
wrote:

Quote:
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org
Date: Monday, June 1, 2009, 7:41 PM


On Mon, Jun 1, 2009 at 3:20 PM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com>
wrote:



Hello the dial plan:



<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>



This i setup from Wikipbx.
What about this in the dialplan?
<action application="set"
data="proxy_media=true"/>
Or alternatively this in the SIP profile?

<param name="inbound-proxy-media"
value="true"/>

I just want to make sure you're actually telling FS
to
Quote:
use proxy media. If I may make a suggestion: use
pastebin.freeswitch.org
Quote:
and pastebin the entire extension in the dialplan as
well as
Quote:
a complete debug log of the call from the FS CLI.
Please see
Quote:
this page for some handy tips on gathering information
for
Quote:
troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC



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anthony.minessale at g...
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PostPosted: Wed Jun 03, 2009 8:41 am    Post subject: [Freeswitch-users] Passthru mode Reply with quote

I am pretty sure inbound-proxy-media forces late-negotation iirc.


On Wed, Jun 3, 2009 at 1:33 AM, Mathieu Rene <mrene_lists@avgs.ca (mrene_lists@avgs.ca)> wrote:
Quote:

On 3-Jun-09, at 2:32 AM, Jim Burke wrote:

Quote:
Fernando,

Try setting 'inbound-late-negotiation' in your SIP Profile.  This will
allow the call to hit the dialplan where you can set proxy_media.
This also assumes you have bypass_media set to false in your dialplan.

Alternatively I beleive you can set "inbound-proxy-media" in the SIP
Profile and this will do the same thing.


But you still need late negotiation for that to work, so in both cases
you need to fix that Very Happy

Math


Quote:


Regards,
Jim

On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL
<fvillarroel@yahoo.com (fvillarroel@yahoo.com)> wrote:
Quote:

Dear,

I can't solve my problem, i was try with:

<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/ubb/$1$2$3"/>

and:

<param name="disable-transcoding" value="true"/> in freeswitch.xml

But receive the same log:

http://pastebin.freeswitch.org/9204

Anyone help me.

Fernando

--- On Mon, 6/1/09, FERNANDO VILLARROEL <fvillarroel@yahoo.com (fvillarroel@yahoo.com)>
wrote:

Quote:
From: FERNANDO VILLARROEL <fvillarroel@yahoo.com (fvillarroel@yahoo.com)>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Monday, June 1, 2009, 8:10 PM

Hello i was try with:

<action application="set" data="bypass_media=true"/>
<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>

This is the log on FS_CLI:

http://pastebin.freeswitch.org/9204

Fernando

--- On Mon, 6/1/09, Michael Collins <msc@freeswitch.org (msc@freeswitch.org)>
wrote:

Quote:
From: Michael Collins <msc@freeswitch.org (msc@freeswitch.org)>
Subject: Re: [Freeswitch-users] Passthru mode
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Monday, June 1, 2009, 7:41 PM


On Mon, Jun 1, 2009 at 3:20 PM,
FERNANDO VILLARROEL <fvillarroel@yahoo.com (fvillarroel@yahoo.com)>
wrote:



Hello the dial plan:



<action application="bridge"
data="sofia/gateway/ubb/$1$2$3"/>



This i setup from Wikipbx.
What about this in the dialplan?
<action application="set"
data="proxy_media=true"/>
Or alternatively this in the SIP profile?

<param name="inbound-proxy-media"
value="true"/>

I just want to make sure you're actually telling FS
to
Quote:
use proxy media. If I may make a suggestion: use
pastebin.freeswitch.org
Quote:
and pastebin the entire extension in the dialplan as
well as
Quote:
a complete debug log of the call from the FS CLI.
Please see
Quote:
this page for some handy tips on gathering information
for
Quote:
troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC



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Anthony Minessale II

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