[asterisk-users] Movistar sip Mexico

[Damian Gonzalez]

Any posible solution?


On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> 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
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- 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
>>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>>>
>>
>>
>>
>> --
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
> --
> _____________________________________________________________________
> -- 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
>



--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131121/adb9ddc4/attachment.html>