[asterisk-users] 20 seconds cut off problem

[Peter Johansson]

Hello. I think i've seen this problem, it was generated by a missing ACK 
on 200 OK. If that is the case try modifying session timer parameters in 
sip.conf so a missing ACK will not lead to call termination.

Peter

Ishfaq Malik wrote:
> Hi
>
> I'm having an issue with just one of the phones (snom300) attached to 
> our asterisk server (1.4.17 using RealTime)
> Sometimes (not consistently), any outbound call cust off at 20 seconds 
> exactly and I see the following in my asterisk console
> [Aug  6 10:37:24] WARNING[1679]: chan_sip.c:1946 retrans_pkt: Maximum 
> retries exceeded on transmission 3c3251e0edaf-4jnjbmy9uupi for seqno 2 
> (Critical Response)
> [Aug  6 10:37:24] WARNING[1679]: chan_sip.c:1970 retrans_pkt: Hanging up 
> call 3c3251e0edaf-4jnjbmy9uupi - no reply to our critical packet.
>
> There as another SIP phone plugged into the same router and that has no 
> issues at all and inbound calls are not affected either.
> The codecs order on the phone match up to those set on the server 
> (g729;alaw;ulaw;;).
>
> There are about 50 other phones attached to the server and none of the 
> others have this issue. Well actually, one did but that person got a new 
> handset (they were previously using a very old and rubbish Grandstream) 
> and the problem immediately stopped.
>
> Has anyone experienced anything like this before?
>
>