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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-ss7

Reply via email to