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 > --
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