[asterisk-ss7] T38 fax issue

[Huseyin Kaya]

Hi,

i set faxopt(gateway)=yes and t38pt_udptl = yes,redundancy,maxdatagram=400 in sip.conf for the peer. 


But still asterisk is sending SIP 200 OK message with G711 and G729 codec in SDP as reply to T38 invite.


Everything will be fine if  itis  possbile to send SIP 488 when asterisk  receives T38 invite.  that is why i tried  t38pt_udptl = no



Best Regards







________________________________
 From: Gregory Massel <greg at csurf.co.za>
To: asterisk-ss7 at lists.digium.com 
Sent: Tuesday, August 26, 2014 9:35 AM
Subject: Re: [asterisk-ss7] T38 fax issue
 


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...
>> 
>> ı 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
> 
>
>
>
>


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