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 <brya...@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" <dgonza...@denwaip.com> > *Sent*: Thursday, November 21, 2013 8:25 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > *Subject*: Re: [asterisk-users] Movistar sip Mexico > > > Any posible solution? > > > On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner <k...@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 >> <dgonza...@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 <al...@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 <dgonza...@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 >> > > > > -- > > > > -- > _____________________________________________________________________ > -- 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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