[asterisk-users] Movistar sip Mexico
Alyed
alyed at vivoxie.com
Thu Nov 21 16:49:22 CST 2013
Have you followed the instructions in:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
??
If possible try with a different ATA since it seems not all of them work
fine with fax pass trough.
Alyed
2013/11/21 Damian Gonzalez <dgonzalez at denwaip.com>
> 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
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> _____________________________________________________________________
>>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com--
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>>>>>>
>>>>>>
>>>>>> --
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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:
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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:
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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:
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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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