See https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway Particularly the line: exten => 1,n,Set(FAXOPT(gateway)=yes) which you need on the SIP side of the DAHDI-SIP bridge On 2014/08/26 08:01 AM, Huseyin Kaya wrote: > Hello > > Does anyone has a working libb7 server that able to handle T38 faxing. > > I am working on this for the last 10 days. But still no solution. > > I am receiving the call from sip and sending to telco with ss7. > > At the beginning of the call everything is fine ( i mean codec > negotiation) . Then the remote sip side detects the fax tone that > remote end sends and Then the sip remote side sends the t38 invite and > asterisk sends SIP 200 OK message with G711 and G729 codec in SDP. > > My customer is saying that if i am sending SIP 200 OK to his T38 > invite .in SDP of SIP 200 OK there should be only T38. > > Also disabling t38 with t38pt_udptl=no didnt change anything. Still > asterisk is sending SIP 200 OK message with G711 and G729 codec in SDP > as reply to T38 invite. > > > I will be glad if current libss7 users can advice me a way to find a > solution on this issue > > Regards > > ------------------------------------------------------------------------ > *From:* Huseyin Kaya <huseyinkaya at yahoo.com> > *To:* Kaloyan Kovachev <kkovachev at varna.net>; > "asterisk-ss7 at lists.digium.com" <asterisk-ss7 at lists.digium.com> > *Sent:* Friday, August 22, 2014 5:32 PM > *Subject:* Re: [asterisk-ss7] T38 fax issue > > Hello > Actually it will be fine If I can send sip 488 not acceptable here to > the re invite of t38 . > But Asterisk is sending sip 200 ok with voice codecs in SDP. > So if i was able to send sip 488 . The fax will contunie with g711 > Setting faxopt(gateway)=yes and t38pt_udptl = > yes,redundancy,maxdatagram=400 didnt work. > By setting these asterisk should send sip 200 with SDP T38 but it is > still sending speech codec(g11u,etc..) in SIP 200 sdp. > Also disabling t38 with t38pt_udptl=no didnt change anything. > > Regards > > > ------------------------------------------------------------------------ > *From: * Kaloyan Kovachev <kkovachev at varna.net>; > *To: * Huseyin Kaya <huseyinkaya at yahoo.com>; > <asterisk-ss7 at lists.digium.com>; > *Subject: * Re: [asterisk-ss7] T38 fax issue > *Sent: * Fri, Aug 22, 2014 1:52:44 PM > > Hi, > in addition to FAXOPT(gateway) you may try to request transmission > medium from telco - see SS7_TMR or SS7_TMR_NUM, as it should be set on > the outgoing (dahdi) channel you need to set it on the SIP channel with > underscore (_SS7_TMR) > > The possible options are defined in libss7.h and SS7_TMR_3K1_AUDIO (or 3 > as num) works fine here. If you can detect the fax calls you may request > 64K_UNRESTRICTED data for them > > On 2014-08-22 15:47, Huseyin Kaya wrote: > > > Hello > > > > I am using Sangoma A104DE with libss7 on Asterisk 11.5.0 and > > terminating calls to telco with ss7 . We are using patched version of > > Libss7 that have timers > > functionality.(https://issues.asterisk.org/jira/browse/SS7-27) > > > > The server is on production for the last 6 months and everything was > > fine > > > > My interconnection with telco is only one way. From sip to ss7 . > > > > Everything is fine except one thing. > > > > One of my customers asked for t38 faxing. then i found myself in > > trouble. > > > > I get several sip traces and found the problem at the end .When i > > receive an invite T38 , asterisk is sending 200 OK message but in SDP > > it is sending g711u,g711a,g729 as codecs. So after this point > > everything is messed up. > > > > I tried to set t38pt_udptl=no in sip.conf , but still asterisk is > > sending sip 200 ok with sdp g711... > > > > ? tried to set t38pt_udptl = yes,redundancy,maxdatagram=400 and > > setvar=FAXOPT(gateway)=yes,20 in sip.conf but still i couldn't manage > > to receive fax > > > > So basically what i need to send fax from sip to telco but could not > > manage to do till now. > > > > Everything except fax is working like a charm. > > > > Is anyone succeed to handle fax with libss7. > > > > I will be glad if an expert can show me a way to achive this. > > > > Best Regards > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20140826/c8b8fa4a/attachment-0001.html>
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