Hello, We have a sip trunk between our voip operator and our asterisk 1.6.2.9
We have no problem during voice communications. But we can not send any t38 fax via this gateway. We tried to trace the error made some tests.. There are 2 main tests we tried to do. As i learned their voip path is like .. we connect to session border controller..then it routes the call to a cisco media gateway if the call is originated thru a pstn/telco line. First test is to send the fax to a client in their SBC device.it was a direct sip2sip fax call. And it succeeded. Then when we tried to fax to a pstn number fax hung up because of communication error. The only error i received was Unknown RTP codec 100 received from xxx.xxx.xxx.xx If i got it right, they say for normal calls they use 99 as dtmf payload. But for fax they use 100. And they asked me if there is a way that i can change dtmf payloads on asterisk or not?? So.. what can i do..or what should i try?? -- _____________________________________________________________________ -- 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
