[asterisk-users] 20 seconds cut off problem

[Ishfaq Malik]

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?

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062