Ok, Folks; I set dtmfmode=info on both contexts. So now no warning message, phone rings, but moment I go off hook, asterisk drops the call, and spit out the following;
-- Executing SetCallerID("SIP/-081341a8", "1001") in new stack -- Executing AbsoluteTimeout("SIP/-081341a8", "6000") in new stack -- Set Absolute Timeout to 6000 -- Executing Dial("SIP/-081341a8", "Sip/[EMAIL PROTECTED]|90|r") in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnect-e3e0 is making progress passing it to SIP/-081341a8 -- SIP/iconnect-e3e0 answered SIP/-081341a8 WARNING[1217602880]: File channel.c, Line 1851 (ast_channel_make_compatible): No path to translate from SIP/-081341a8(4) to SIP/iconnect-e3e0(256) WARNING[1217602880]: File app_dial.c, Line 672 (dial_exec): Had to drop call bec ause I couldn't make SIP/-081341a8 compatible with SIP/iconnect-e3e0 == Spawn extension (vobb-in, 81510xxxxxx, 3) exited non-zero on 'SIP/-081341a 8' -- Executing Hangup("SIP/-081341a8", "") in new stack == Spawn extension (vobb-in, h, 1) exited non-zero on 'SIP/-081341a8' It seems like * is trying to translate the codec. I have set G729 for both contexts. I thought if there is no codec translation, asterisk can handle pass through. Cheers Sathya > -----Original Message----- > From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 19, 2003 11:01 AM > To: Eric Wieling; [EMAIL PROTECTED] Digium. Com > Subject: RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning > message > > > I am sorry I mean dtmfmode=info > > > -----Original Message----- > > From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, November 19, 2003 10:34 AM > > To: Eric Wieling; [EMAIL PROTECTED] Digium. Com > > Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message > > > > > > Hi, > > > > Thanks Jeramy and Eric. > > > > Sorry for my ignorance. I still did not get the point. > > > > Do you mean that I have to set each of my context in sip.conf > > with dtmfmode=inband ? > > > > I have the GS phone set as DTMF mode = Via SIP Info. Would that > > need to be change to something else ? > > > > (Send DTMF: in-audio via RTP (RFC2833) via SIP INFO) > > > > Cheers > > > > Sathya > > > > > > Date: Wed, 19 Nov 2003 06:15:35 -0600 > > From: Eric Wieling <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message > > Reply-To: [EMAIL PROTECTED] > > > > Jeremy McNamara wrote: > > > > > Don't try to do inland DTMF on anything but G.711. > > > > > > Jeremy McNamara > > > > > > > Someone really needs to patch Asterisk to print some ugly warning or > > notice to the Asterisk console when the codec that is being used for a > > call is not ulaw/alaw and trhe dtmfmode=inband (manyually or > > automagically set) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users