[asterisk-users] Asterisk Avaya SIP Trunking One Way Audio

Shariq Khan shariqrazakhan at gmail.com
Fri Apr 8 05:05:30 CDT 2011


Yes, disabling reinvites solved the problem :)

Thanks.

--
Regards,
Shariq Khan
0333-3501125



On Fri, Apr 8, 2011 at 6:38 AM, Lyle Giese <lyle at lcrcomputer.net> wrote:

> On 04/07/11 03:00, Shariq Khan wrote:
>
>> I am facing one way audio problem in sip trunking between asterisk and
>> avaya.
>>
>>               +-------------+       +----+
>>               | avaya sip   |-------| P1 |
>>               +-------------+       +----+
>>                      |
>>                      |
>>                      |
>>               +-------------+
>>               |  Asterisk   |               WAN
>> -------------------------------------------------
>>               |             |               LAN
>>               +-------------+
>>                  |
>>                  /
>>        +----+   /
>>        | P2 |--+
>>        +----+
>>
>> When P1 dial P2, P2 hears voice clear but P1 could not hear any voice.
>>
>> My sip.conf is
>>
>> [avaya]
>> type=peer
>> fromdomain=xx.xx.xx.xx
>> host=xx.xx.xx.xx
>> disallow=all
>> allow=ulaw
>> dtmfmode=rfc2833
>> canreinvite=yes
>>
>>
>> --
>> Regards,
>> Shariq Khan
>> 0333-3501125
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
> Turn off reinvite on all extensions and SIP trunks involved and try again.
>
> Lyle Giese
> LCR Computer Services, Inc.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
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