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 <dgonza...@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 <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 >
-- _____________________________________________________________________ -- 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