post a trace of FS after pressing f8 from the cli detailing the entire call and we can have a look.
On Fri, Sep 19, 2008 at 6:29 PM, Matt Darnell <[EMAIL PROTECTED]> wrote: > On Wed, Sep 17, 2008 at 10:14 AM, Matt Darnell <[EMAIL PROTECTED]> > wrote: > > On Fri, Jul 25, 2008 at 8:51 PM, UV <[EMAIL PROTECTED]> wrote: > >> Yes I did, but you might not even need that. > >> Try adding <param name="pass-rfc2833" value="true"/> in your external > SIP > >> profile and see if it solves the problem. > >> > > > > I am still trying to get the DTMF 100%, I added the value but get this > > message in the debug log: > > [WARNING] mod_sofia.c:787 sofia_receive_message() Cannot pass 2833 on > > a transcoded call > > > > It does not appear that any transcoding is happening from the SIP > > setup messages. > > Steve, > > Thanks for the response, here is the architecture: > > SIP Provider <-> SIP UDP <-> Internet <-> Freeswitch <-> LAN <-> SIP > TCP <-> Exch 2007 > > It appears to be G711 throughout the entire call, is it transcoded > just because Freeswitch is bridging the call? > > -Matt > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400
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