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I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok.
But when I start calling on a DID on Asterisk A then the call is being routed to Asterisk 'B' and After 38 seconds call has been disconnected showing following warnings :

Retransmission timeout reached on transmission 11bc71e029119e5877806ed40fcde691@111.xxx.xxx.xxx:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Hanging up call 11bc71e029119e5877806ed40fcde691@111.xxx.xxx.xx:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Any ideas ?

lmo
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Vivek Raj
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  • can you call the DDI on your server 'A' without the trunk (A-B) configuation? To make sure your dialplan is correct. – pce Feb 28 '14 at 12:47
  • Yes, Its working perfectly without trunk configuration. The main issue is call is being disconnected after 38 seconds, Before 38 seconds I can listen audio files to, As soon as the call reaches to 38th seconds , I t dropped – Vivek Raj Feb 28 '14 at 13:45
  • so...have you solved your problem yet? – Riad Nov 20 '14 at 08:40
  • Yes, As I mentioned it below it was due to nat setting in sip.conf – Vivek Raj Dec 04 '14 at 11:52
  • please tag it as solved ;) thanks – lmo Dec 09 '14 at 08:51

5 Answers5

8

By default Asterisk sends a RE-INVITE request after a call is established.

But most sip clients and sip servers in the market do not accept RE-INVITE requests. For this reason, when Asterisk sends a RE-INVITE after a call is established, the other side does not answer the request. So, after 32 seconds, Asterisk hangs up the call.

To solve the problem, you need to disable the RE-INVITE feature of Asterisk if your client software does not accept RE-INVITE requests. To do this, you need to edit the sip.conf file in Asterisk to include:

canreinvite = no
Jon Egeland
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Iftekhairul Alam
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4

Such situation can be spot when you have nat issues or firewall issue

See this article http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

For more info you can enable sip debug by using

 asterisk -r
 sip set debug on
arheops
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  • I have removed the firewall settings, This issue is not related firewall, For Nat I have used nat=force_rport,comedia in sip.conf. – Vivek Raj Feb 28 '14 at 14:25
  • From debug log, Asterisk server 'B' keeps on showing Retransmitting #{1-9}(NAT), where {1-9} showing retransmition attempt, – Vivek Raj Mar 01 '14 at 07:07
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    Retransmitting #3 (NAT) to 111.XXX.XXX.X:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.XXX.XXX.X:5060;branch=z9hG4bK5e38e44b;received=111.XXX.XXX.X;rport=5060 From: ;tag=as086c55da To: ;tag=as36de7a5b Call-ID: 72cd617b6e36fd187315d5fd5d892e5d@111.XXX.XXX.X:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp – Vivek Raj Mar 01 '14 at 07:08
  • So you have nat or firewall issues like i say. – arheops Mar 01 '14 at 16:33
  • Thanks for your reaponse, please suggest me the solution to overcome this problem – Vivek Raj Mar 02 '14 at 17:20
  • Setup nat/fix firewall. – arheops Mar 02 '14 at 20:15
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These incidents usually associated with NAT problems.

If you're sure that this isn't your problem, take a look at router configuration. Some routers are configured by default with "SIP ALG" option.

In some cases, this option should be off because implementation is incomplete.

Try it, and let me known if it works properly.

Sisco Casasempere
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    Thanks for your suggestion. Actually I figured it out earlier, It was due to NAT settings. I had resolved this issue. – Vivek Raj Aug 30 '14 at 20:04
0

make sure you have correct ip address in 'externip=' in sip.conf under /etc/asterisk.

sofs1
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0

Sounds like nat problems. Can you share your sip configs so we can take a look?

Have you set your extenip and localip?

Wesley Schravendijk
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