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 <al...@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 <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 >> > > > -- > _____________________________________________________________________ > -- 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