[asterisk-users] Movistar sip Mexico

Damian Gonzalez dgonzalez at denwaip.com
Thu Nov 21 16:13:10 CST 2013


Hi,

I have Asterisk 10.12.1. I can not figure out the solution.

Thank you for your help.

Best Regards


On Thu, Nov 21, 2013 at 7:07 PM, Alyed <alyed at vivoxie.com> wrote:

> 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
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
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>>>>>
>>>>>
>>>>> --
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>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>> --
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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
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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
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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