[asterisk-ss7] T38 fax issue

Huseyin Kaya huseyinkaya at yahoo.com
Tue Aug 26 01:01:07 CDT 2014


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



 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


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. 


 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 

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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