[Asterisk-Users] g723 to g723 SIP call - warning message

[Sathya Weerasooriya]

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/1510xxxxxx at iconnectx|90|r")
 in new stack
    -- Called 1510xxxxxx at iconnect
    -- 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:sathyaw at sbcglobal.net]
> Sent: Wednesday, November 19, 2003 11:01 AM
> To: Eric Wieling; Asterisk-Users at Lists. 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:sathyaw at sbcglobal.net]
> > Sent: Wednesday, November 19, 2003 10:34 AM
> > To: Eric Wieling; Asterisk-Users at Lists. 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 <eric at fnords.org>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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)