Asterisk,SIP Retransmission timeout

Question

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

Answer 1

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.

Answer 2

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

Answer 3

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

Answer 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

Answer 5

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

Have you set your extenip and localip?