Asterisk,SIP Retransmission timeout

11
votes

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 [email protected]: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 [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Any ideas ?

5
sipasterisk
can you call the DDI on your server 'A' without the trunk (A-B) configuation? To make sure your dialplan is correct. – pce
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
so...have you solved your problem yet? – Riad
Yes, As I mentioned it below it was due to nat setting in sip.conf – Vivek Raj
please tag it as solved ;) thanks – lmo

5 Answers

9
votes

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
4
votes

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
2
votes

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.

0
votes

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

0
votes

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

Have you set your extenip and localip?