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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 [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 ?
481
Other SIP requests are only based on request - reply. There's no ACK, no three-way handshake. In Asterisk we mark some of these as CRITICAL - they need to go through for the call to work as expected. Some are non-critical, we don't really care what happens with them, the call will go on happily regardless.
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.
In order to make sure that the whole call setup sequence works and that we have a call, a SIP client retransmits messages if there's too much delay between request and expected response. We retransmit a number of times while waiting for the first response. We retransmit the answer to an incoming INVITE while waiting for an ACK.
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:
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
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
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.
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