[asterisk-users] Movistar sip Mexico

[Kristian Kielhofner]

It is possible that Asterisk requires an rtpmap even for static payload
types (I'm not sure about this).  The INVITE from your provider omits
rtpmap for payload type 18 (G729) which is perfectly valid.


On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez <dgonzalez at denwaip.com>wrote:

> Hello,
>
> Thanks for the quickly response. I have only G729 in the peer but I have
> t38pt_udptl= yes
>
> If I put t38pt_udptl=no , asterisk reject the call with 488 code.
>
> The problem is that Movistar send T38 codec in all calls and I need ignore
> only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38
> I have to negociate a fax call.
>
> Thanks.
>
>
> On Wed, Nov 20, 2013 at 4:46 PM, Alyed <alyed at vivoxie.com> wrote:
>
>> Think you only need to make sure you have in your sip.conf file these
>> configs:
>>
>> [your-device-name]
>> .....
>> .....
>> disallow=all
>> allow=g729
>> .....
>> .....
>>
>>
>> Alyed
>>
>> 2013/11/20 Damian Gonzalez <dgonzalez at denwaip.com>
>>
>>> Hello,
>>>
>>> I have a problem with movistar in Mexico with a sip calls. Movistar send
>>> to me T38 and G729 in the INVITE and they say that I have to ignore T38 and
>>> use G729 in the voice call.
>>>
>>> When a fax call is made Movistar send only T38 in the INVITE.
>>>
>>> Invite example:
>>>
>>> v=0
>>> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
>>> s=sip call
>>> c=IN IP4 192.168.1.2
>>> t=0 0
>>> m=audio 6370 RTP/AVP 18 101
>>> a=fmtp:18 annexb=yes
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20
>>> m=image 6372 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38FaxMaxBuffer:1100
>>> a=T38FaxMaxDatagram:612
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> How can I  ignore T38 and use only G729 for this call?.
>>>
>>> Thanks for your help.
>>>
>>> Damian
>>>
>>>
>>> --
>>>
>>>
>>> --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
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>
>
>
> --
>
>
> --
> _____________________________________________________________________
> -- 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
>



-- 
Kristian Kielhofner
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