[asterisk-users] 20 seconds cut off problem

[Ishfaq Malik]

Cheers for that.

We solved the problem this time by updating the routers firmware to the 
latest version but if it happens again I'll look into what you're saying.
Can I just clarify that you mean the RTP Timers though? It's a 
production server and doing sip reloads is massively frowned upon.

Thanks again

Ish

Peter Johansson wrote:
> 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?
>>
>>   
>>     
>
>
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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062