[asterisk-users] chan_sip.c:3641 retrans_pkt: Retransmission timeout

[Benoit Panizzon]

Hello

We have a Genband C3 Switch and a couple of customers that operate asterisk 
PBXes connected via SIP Trunk. All of them still use some 1.6.X asterisk and 
this works fine.

One customer uses a 1.8 version and has a very strange problem:

Asterisk 1.8.10.0-1digium1~lucid built by pbuilder @ nighthawk on a x86_64 
running Linux on 2012-03-06 01:51:21 UTC

When a call reaches this asterisk the call get's dropped after a couple of 
seconds with this message:

[Mar 27 13:47:35] WARNING[29806]: chan_sip.c:3641 retrans_pkt: Retransmission 
timeout reached on transmission F47F6F67 at 7f33ff47 for seqno 907689273 
(Critical Response) -- See 
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Mar 27 13:47:35] WARNING[29806]: chan_sip.c:3670 retrans_pkt: Hanging up call 
F47F6F67 at 7f33ff47 - no reply to our critical packet (see 
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Well, the Wiki is not very helpfull, as the problems explained there do not 
seem to be the cause.

I sniffed the connection on the Asterisk Box itself and what I see is this 
flow of SIP messages:

C3 <=> Asterisk

=> Invite +sdp
<= 401 Unauthorized
=> Ack
=> Invite +sdp +auth
<= 100 Trying
<= 180 Ringing
<= RTP Audio
<= 200 OK +sdp
=> Ack
<=> RTP Audio
<= 200 OK +sdp
=> Ack
<= 200 OK +sdp
=> Ack
<= 200 OK +sdp
=> Ack
<= Bye
=> Ack
<= Bye
=> Ack
<= Bye
=> Ack

Well apparently the C3, from my point of view sends correct Acks to the 
Asterisk, but somehow the Asterisk ignores them.

I did try to get more information with debug and verbose level set to '99', 
but I don't see more messages

Does anyone have a clue, why acks could be not accepted by asterisk 1.8.10.0 ?

The other way round (asterisk => c3) the calls work fine.

Regards

Benoit Panizzon
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