[asterisk-users] Asterisk hangup all incoming calls after 10 seconds

[Bruno Camargo]

Toooooo early... call droped after 11 seconds now... different log.

[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: SIP TIMER: Rescheduling
retransmission #13781 (6) SIP/2.0 - 1
[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: ** SIP timers: Rescheduling
retransmission 7 to 4000 ms (t1 100 ms (Retrans id #13781))
[Mar 17 22:19:20] DEBUG[4459] rtp.c: Got RTCP report of 176 bytes
[Mar 17 22:19:21] WARNING[2783] chan_sip.c: Maximum retries exceeded on
transmission 6b24c4992c31e3b264e271ba29f33db4 at 200.229.195.226 for seqno 102
(Critical Response)
[Mar 17 22:19:21] DEBUG[2783] chan_sip.c: Setting SIP_ALREADYGONE on dialog
6b24c4992c31e3b264e271ba29f33db4 at 200.229.195.226
[Mar 17 22:19:21] WARNING[2783] chan_sip.c: Hanging up call
6b24c4992c31e3b264e271ba29f33db4 at 200.229.195.226 - no reply to our critical
packet.
[Mar 17 22:19:21] DEBUG[4459] channel.c: Didn't get a frame from channel:
SIP/7977529-081931c0
[Mar 17 22:19:21] DEBUG[4459] channel.c: Bridge stops bridging channels
SIP/7977529-081931c0 and SIP/242-081910e8
[Mar 17 22:19:21] DEBUG[4459] channel.c: Hanging up channel
'SIP/242-081910e8'
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Hangup call SIP/242-081910e8, SIP
callid 6fa30fba46a0cd93213baeab361530b8 at 192.168.20.249)
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Strict routing enforced for
session 6fa30fba46a0cd93213baeab361530b8 at 192.168.20.249
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: *** SIP TIMER: Initializing
retransmit timer on packet: Id  #-1

and not a clue.

THanks alot

On Wed, Mar 17, 2010 at 10:05 PM, Bruno Camargo <mustardahc at gmail.com>wrote:

> Hello Gentleman,
>
> I guess the problem was the codec.
>
> I have allowed only g711u for testing purposes and the incoming call
> endured for 1 minute, until the caller hanged.
>
> Thanks a lot for the support.... but there are tons of questions yet to be
> answered!
>
> Thanks
>
>
> On Wed, Mar 17, 2010 at 2:12 PM, Fred Posner <fred at teamforrest.com> wrote:
>
>> On Mar 17, 2010, at 1:05 PM, Bruno Camargo wrote:
>>
>> > Hi Giorgio,
>> >
>> > So it means that Asterisk has no native support for g729 ?
>> >
>> > Thanks
>> >
>> > --
>> > BrCaBadT
>> > --
>>
>> Depends on your definition of support. It supports passthrough... but if
>> you're using it locally on a bridge on transcoding, you'll need to purchase
>> a license. The codec itself is non-G729 compliant.
>>
>>
>> ---fred
>> http://qxork.com
>>
>>
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
>
> --
> BrCaBadT
>



-- 
BrCaBadT
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