[asterisk-users] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly

[Miguel Oyarzo]

It seems a firewall or signaling problem. The calling part is not 
sending a ACK response to your host because it never get an "OK 200" 
from your host.

In other words, the called part is trying to send to the calling part
a) TRYING 100, then
b) RING 180 and  finally
c) OK 200

but the calling part seems not being receiving no signals from you host.
As a result, your host has sent 4 times "SIP/2.0 200 OK" 
(retransmissions) to the calling part but it never got an ACK from the 
other end to establish the communication.
Then, the link is destroyed.

regards,

-- 
==================================
Miguel Oyarzo
Senior [ Network | Systems Design ] Engineer
http://www.linkedin.com/in/mikeaustralia
Linux User: # 483188 - counter.li.org
Melbourne, Australia




On 11/11/2012 9:46 PM, Eric Kuhnke wrote:
> Hi all,
>
>
> I'm trying to troubleshoot an issue with my SIP service.  All outgoing
> calls work normally.  The following is a SIP debug log from Asterisk.  The
> test setup is as follows:
>
> One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk
> to my local FreePBX/Asterisk 11.0 server which is at 10.0.0.17.
>
> The Yealink phone doesn't seem to have any problem placing outgoing calls
> through the Asterisk server, which is registered to Diamondcard.  I can
> reach both the Asterisk server itself (for example to use voicemail) or
> call any number on the PSTN.  Likewise I have the server configured to pass
> incoming DID calls for myDIDnumber to extension 10.  Calls from the PSTN to
> myDIDnumber ring the phone, including CID passing, and will connect a full
> duplex audio call session.  The problem is that the phone won't stay
> connected longer than 13 to 17 seconds.
>
> When the phone is manually configured to use my account and password on the
> diamondcard servers directly, both incoming and outgoing calls work
> normally, with RTP/UDP port 5060 traffic passing through my NAT without
> trouble.  I have made no special modifications to the NAT.
>
> 13 seconds after picking up an incoming call, the phone disconnects at the
> same time as the log shows this:
>
> [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Retransmission timeout
> reached on transmission08a728706baea3b740aa806e41e9d13d at 69.71.222.196  <mailto:08a728706baea3b740aa806e41e9d13d at 69.71.222.196>  for
> seqno 103 (Critical Response) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 17853ms with no response
> [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Hanging up call
> 08a728706baea3b740aa806e41e9d13d at 69.71.222.196  <mailto:08a728706baea3b740aa806e41e9d13d at 69.71.222.196>  - no reply to our critical
> packet (seehttps://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>
>    
>    
> *The full log and configuration is at:*
> *http://pastebin.com/1Mgn72vN*
>
>
> --
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