[asterisk-users] Movistar sip Mexico

Alyed alyed at vivoxie.com
Thu Nov 21 16:07:33 CST 2013


Which version of Asterisk are you using?

According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless you
are using Asterisk 10, there's quite some patching (or buying) you'll need
to be doing.

Alyed


2013/11/21 Bryant Zimmerman <BryantZ at zktech.com>

> Can you funnel them through a specific inbound dial context. Then force a
> re-invite to g729?
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> ------------------------------
> *From*: "Damian Gonzalez" <dgonzalez at denwaip.com>
> *Sent*: Thursday, November 21, 2013 8:25 AM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users at lists.digium.com>
> *Subject*: Re: [asterisk-users] Movistar sip Mexico
>
>
> 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 --
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>>>>>
>>>>
>>>>
>>>> --
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>>>
>>>
>>>
>>> --
>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>
>>
>>
>> --
>> 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
>>
>
>
>
> --
>
>
>
> --
> _____________________________________________________________________
> -- 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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