VoIP Mailing List Archives

[andregronwald78 at gma...]

Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but I'm unsure.

A registration to Sipgate is established successfully:


<Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 pjsip_sipgate/sip:sipgate.de:5060                       pjsip_sipgate     Registered     


Calling the registered number is even successfully shown in asterisk (it is a freepbx installation).
But when doing a second call the number is busy ("provider" busy, I don't see anything in asterisk verbose mode).
Sending a pjsip unregister results in the following messages:

[2016-10-15 10:03:22] WARNING[10162]: res_pjsip_outbound_registration.c:761 schedule_retry: No response received from 'sip:sipgate.de:5060' on registration attempt to 'sip:2636146e0@sipgate.de:5060 ([email]sip:2636146e0@sipgate.de:5060[/email])', retrying in '60'
    -- Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Reachable.  RTT: 434.393 msec
  == Endpoint pjsip_sipgate is now Reachable

so it is somewhat clear, why i get a busy, because the endpoint is not reachable. But WHY is the endpoint not reachable?

Regarding the architecture: I have two routers cascaded, that is unfortunately necessary. On the first router (vDSL-access router) I have forwarded nearly everything to the second router (Bintec rj 353), where a port forwarding for relevant ports (sip and pjsip (udp and tcp), rtp (udp)) is configured. IF a call goes through, nearly everything is working (audio only incoming, but that is another issue).

STUN is configured. FreePBX Firewall is disabled.

Kind regards,
andre

[andregronwald78 at gma...]

Very interesting: I have another provider configured, that was not reachable as well. I disabled the STUN-server (external STUN server), and now the second registration works fine, BUT with the same "error" messages (unreachable etc) as the other provider. But in contrast the number is always reachable!!!

Is there any explanation for this? I just want to understand... ... and solve it.

regards,
andre

Am 15.10.2016 um 10:11 schrieb Andre Gronwald:

[lardconcepts at gmail.com]

All other things aside, this stands out immediately:

RTT: 434.393 msec

That's almost half a second round trip for a packet. I'm amazed
anything works at all. For SIP connections, mine are usually about
26ms max, anything above about 35 is bad. Looks like a serious config
issue.

Try pinging and see what you get - my ping times to sipgate.de from
the UK are Best:13.6ms Worst 13.8ms across 100 pings.

I could be wrong, but I'd be surprised if that wasn't causing
problems, at least with audio.


On 15 October 2016 at 09:11, Andre Gronwald <andregronwald78@gmail.com> wrote:

[andregronwald78 at gma...]

ping times are fine as well:

[root@freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57 time=46.8 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=5 ttl=57 time=47.1 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=6 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=7 ttl=57 time=47.1 ms
^C
--- sipgate.de ping statistics ---
7 packets transmitted, 7 received, 0% packet loss, time 6360ms
rtt min/avg/max/mdev = 46.406/46.809/47.191/0.393 ms
[root@freepbx asterisk]#


this high RTT appears only sometimes.  After removing STUN-server it looks better, did two test calls right now, both gone through immediately. At the end of the second test call I see:

    -- Executing [s@app-announcement-1:5] Playback("PJSIP/pjsip_sipgate-00000003", "custom/araz01&custom/07-polly,noanswer") in new stack
    -- <PJSIP/pjsip_sipgate-00000003> Playing 'custom/araz01.alaw' (language 'en')
    -- Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Reachable.  RTT: 493.094 msec
  == Endpoint pjsip_sipgate is now Reachable
    -- <PJSIP/pjsip_sipgate-00000003> Playing 'custom/07-polly.slin' (language 'en')
    -- Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Unreachable.  RTT: 0.000 msec
  == Endpoint pjsip_sipgate is now Unreachable


Why do I have that loss of registrations?

here my pjsip config for sipgate.de:

freepbx*CLI> pjsip show registration pjsip_sipgate

 <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 pjsip_sipgate/sip:sipgate.de:5060                       pjsip_sipgate     Registered     

 ParameterName            : ParameterValue
 ========================================================
 auth_rejection_permanent : true
 client_uri               : sip:2636146e0@sipgate.de:5060 ([email]sip:2636146e0@sipgate.de:5060[/email])
 contact_user             : 2636146e0
 endpoint                 :
 expiration               : 600
 fatal_retry_interval     : 0
 forbidden_retry_interval : 0
 line                     : false
 max_retries              : 10
 outbound_auth            : pjsip_sipgate
 outbound_proxy           :
 retry_interval           : 60
 server_uri               : sip:sipgate.de:5060
 support_path             : false
 transport                : 0.0.0.0-udp

Remind: Endpoint is currently unreachable, but asterisk shows "Registered". Test call fails at this moment.


regards,
andre

[lardconcepts at gmail.com]

Hmmm, sorry, I can't think of anything except... why do you need the
STUN server? And are you sure that all the ports in your router
definitely match the ones Asterisk thinks it's using?

Then there is always the SIP-ALG problem with some routers, which some
people have been able to overcome by switching to TLS, and I see that
SIPgate offer TLS.
You could try making a free certificate and going TLS which uses port
5061. No promises, but worth a try as it fixed the issue for a
different poster.

The only other thing I can find while Googling for this, which solved
it for someone else, was related to DNS server issues, but this seems
unlikely (although not impossible).

On 15 October 2016 at 10:07, Andre Gronwald <andregronwald78@gmail.com> wrote:

[andregronwald78 at gma...]

Thanks Jonathan for your support.

I would like to avoid TLS at the moment  (in general I am a fan of secured communication!) because the other provider is not supporting TLS. And sipgate is just used for testing.

However I can see the following which is quite interesting:

[2016-10-15 11:20:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Reachable.  RTT: 433.814 msec
[2016-10-15 11:20:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:21:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Unreachable.  RTT: 0.000 msec
[2016-10-15 11:21:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Endpoint pjsip_sipgate is now Unreachable
[2016-10-15 11:30:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Reachable.  RTT: 439.006 msec
[2016-10-15 11:30:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:31:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Unreachable.  RTT: 0.000 msec
[2016-10-15 11:31:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Endpoint pjsip_sipgate is now Unreachable
[2016-10-15 11:40:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Reachable.  RTT: 433.426 msec
[2016-10-15 11:40:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:41:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Contact pjsip_sipgate/sip:2636146e0@sipgate.de:5060 ([email]pjsip_sipgate/sip:2636146e0@sipgate.de:5060[/email]) is now Unreachable.  RTT: 0.000 msec
[2016-10-15 11:41:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: Endpoint pjsip_sipgate is now Unreachable

I think that the times are matching exactly the qualify frequency and registry expiration - expiration is set to 600s, and qualify frequency to 50s. Seems that the qualify requests are not supported (this is the case for the other provider as well!). So maybe I should work without sip qualify.

Besides this I have another curiousity:
One call:
    -- Executing [s@app-announcement-1:3] Wait("PJSIP/pjsip_sipgate-00000019", "1") in new stack
       > 0x7fabf004bfd0 -- Probation passed - setting RTP source address to 217.10.77.109:16248
Another call:
    -- Executing [s@app-announcement-1:3] Wait("PJSIP/pjsip_sipgate-0000001a", "1") in new stack
       > 0x7fabf0070bb0 -- Probation passed - setting RTP source address to 192.168.2.1:7074

??? 217.10.77.109 is sipgate.de -> ok. 192.168.2.1 is my vDSL-access-router ??? Why does the RTP source address changes? that must not happen.


And another observation: I am registered to sipgate.de, fine. Incoming call is processed,  announcement is played. But when the caller hangs up asterisk is not recognizing it. it takes about 16s until the channel is closed after hangup?


regards,
andre

[andregronwald78 at gma...]

hi,
let me explain in detail, what i have configured and what is happening now:

1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device
192.168.3.99 (same ports)

other direction is totally open.

I observed that all sip calls are closed exactly after 32s. call is
disconnected on calling side as well... seems to be a timeout issue.

here i have some debug logs. I see lot of requests from asterisk to
sipgate.de, which are not answered. but communication is going fine in
both directions (otherwise registration would not be possible?):


<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "02363361779" <sip:02363361779@sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0@sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de
CSeq: 103 INVITE
Contact: <sip:0xxxxxxxx9@217.10.77.115:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Content-Length: 394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9@sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0@sipgate.de>
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length: 0

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9@sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "0xxxxxxxx9" <sip:0xxxxxxxx9@sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0@sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de
CSeq: 103 INVITE
Contact: <sip:0xxxxxxxx9@217.10.77.115:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Content-Length: 394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9@sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9@sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Executing [s@app-announcement-1:3]
Wait("PJSIP/pjsip_sipgate-00000003", "1") in new stack

[ian.gilmour.x at gmail...]

Hi,

I don’t see any SIP ACK’s in your trace.

Is the SIP 200 OK reaching the originating caller, or being blocked on
the way through?

Asterisk will tear down the call after ~30secs of audio playing in both
directions if it doesn't receive the SIP ACK.

Regards,

Ian


On 15/10/2016 12:05, Andre Gronwald wrote:

[andregronwald78 at gma...]

ok, solved the firewall issue.
A first test call worked fine. Another one now still gets disconnected
after 32s.

But in FW there are no blocked packets anymore?!

And I don't understand why the registration to the same IP and same Port
is working, but not later transmission of further SIP packets? that
doesn't sound logical to me. What do you think?

regards,
andre

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[edilsonamaral at yahoo...]

Have you tried setting keepalive(20 seconds) on your sip.conf and on your phones ?







From: Andre Gronwald <andregronwald78@gmail.com>
To: asterisk-users@lists.digium.com
Sent: Saturday, October 15, 2016 9:17 AM
Subject: Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore


ok, solved the firewall issue.
A first test call worked fine. Another one now still gets disconnected
after 32s.

But in FW there are no blocked packets anymore?!

And I don't understand why the registration to the same IP and same Port
is working, but not later transmission of further SIP packets? that
doesn't sound logical to me. What do you think?

regards,
andre

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com--

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[andregronwald78 at gma...]

ok, now it is getting weird...
actually i don't see any firewall issues, but i am not able to get a call from outside to my sipgate account. in asterisk nothing is visible, core set verbose is activated.
sngrep (on my asterisk server) shows me indeed the invite from sipgate!?

What I see via sngrep is the following options-flow:
192.168.3.50:55060 -> 217.10.79.9:5060
217.10.79.9:5060 -> 192.168.3.50:48757 (200 OK)

shouldn't sipgate answer on the same port that the communication initiated??? in this case 55060?