I have been making calls from X-pro through oh323 channel to PSTN. Below are details: extension.conf ---------------------- exten => _00X.,1,Dial(OH323/${EXTEN},60)
sip.conf ---------------- [general] ;progressinband=never [test1] dtmfmode=rfc2833 disallow=all allow=g729 -- Executing Dial("SIP/1000-a071", "OH323/001234|60") in new stack -- H.323 call to 001234 with codec(s) g729 -- Called 001234 -- OH323/L17774 is ringing -- OH323/L17774 answered SIP/1000-a071 -- Hungup 'OH323/L17774' Thanks for help. Newbie On Apr 8, 2005 12:39 AM, Chris Modesitt <[EMAIL PROTECTED]> wrote: > > Hi all, > > > Did I make my issue clear? Can any one give me a big hand? > > > Many thanks. > > Newbie > > > On Apr 5, 2005 12:59 AM, VoIP Newbie <[EMAIL PROTECTED]> wrote: > > Hi all, > > > > When I made calls from SIP phones through a analog PSTN gateway to > > PSTN phones, I could hear rings twice on my SIP phones. From my best > > guess, the first one from * and the second one from analog PSTN line. > > Am I right? Is it configuration related? Can the two rings be reduced > > to a single ring like we make normal analog PSTN call from a normal > > PSTN phone? > > > > Many Thanks. > > > > Newbie > > I will need to know a little more about your setup, but here are a couple of > pointers. > > If you have a ",r" in your dial plan asterisk will fake ring back aka > exten => 1000,1,Dial(SIP/1000,300,r) > > To go any further we will need to know your IP phone's make and model, how > you are terminating to the PSTN and your DTMF mode. > > > Chris > > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users