Dear David, Please find on http://pastebin.com/m69b8559d my sip.conf file
Thanks a lot On Fri, Feb 27, 2009 at 1:05 PM, David fire <ddf...@gmail.com> wrote: > paste your sip.conf. > David > > 2009/2/26 michel freiha <mich...@gmail.com> > >> Dear All, >> I have created an inbound context in SIP .conf that forward incoming call >> to opensips server...The problem appears as soon as I enable t38pt_udptl = >> yes...The Asterisk negotiate the SIP session with OpenSIPS without adding >> voice codec to INVITE packet...It just contains T.38 protocol...When >> t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with >> OpenSIPS and cal success..Any suggestion here? >> >> Thanks >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > (\__/) > (='.'=)This is Bunny. Copy and paste bunny into your > (")_(")signature to help him gain world domination. > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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