Many thanks for clarification, Dan ----- Original Message ----- From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, June 01, 2003 5:35 AM Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?
> > Because your original question asked about G.723, which Asterisk > cannot transcode because there is no codec support for it within > Asterisk. > > Summary: If all you want to do is have Asterisk "relay" the RTP > (voice) data between two endpoints, pretty much any codec can be > used, since Asterisk doesn't have to interpret the data stream for > any reason - it's just moving the data around, and not "listening" or > "talking" in the stream. However, if you want to do something clever > with that data/sound stream, such as listening for the "#" key (the > "t" option in a Dial statement) then you'll need to be using a codec > that Asterisk understands (G711, gsm, iLIBC, etc.) > > JT > > > >I'm not sure that I understand you. > >Why not to do transcoding if sometimes required? > > > >Thanks, > >Dan > >----- Original Message ----- > >From: "John Todd" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Saturday, May 31, 2003 7:35 PM > >Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not? > > > > > >> There is one more note: make sure you don't have any options in your > >> Dial statement that require the Asterisk server to do transcoding. > >> Such options would be "r", or "m", or "t", which will cause Asterisk > >> to need to listen and/or insert sounds in an audio stream if I > >> understand previous conversations here to be correct. I would just > >> remove all options from your Dial statments entirely and see what you > >> get. > >> > >> JT > >> > >> > >> >On Sat, 2003-05-31 at 10:51, Dan wrote: > >> >> Hi, > >> >> > if you turn off the reinvite in the asterisk configs for those > >ata186s > >> >> > then it will pass through asterisk even if asterisk doesn't > >understand > >> >> > the codec. > >> >> So I must have: > >> >> canreinvite = no > >> >> in sip.conf file for the specific phone? > >> > > >> >yes > >> > > >> >> Then the call is passed through Asterisk without any conversion? > >> > > >> >yes > >> > > >> >> How can I do to pass all the calls through Asterisk, even if a codec > >> >> conversion is required or not? > >> > > >> >canreinvite=no > >> >The whole point is you don't reinvite the phones to talk to each other > >> >instead of passing through asterisk. > >> > > >> >> ----- Original Message ----- > >> >> From: "Steven Critchfield" <[EMAIL PROTECTED]> > >> >> To: <[EMAIL PROTECTED]> > >> >> Sent: Saturday, May 31, 2003 5:27 PM > >> >> Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or > >not? > >> >> > >> >> > >> >> > On Sat, 2003-05-31 at 08:06, Dan wrote: > >> >> > > Hi all, > >> >> > > > >> >> > > One short question. > >> >> > > When one extension (let's say ATA-186, SIP image, G.723 codec > >> >> > > selected) try to call an external SIP address like: > >> >> > > SIP/[EMAIL PROTECTED], where another identical ATA-186 is available > >with > >> >> > > G.723 codec selectrd, > >> >> > > after the signaling phase, the call is established through > >Asterisk or > >> >> > > directly between the two ATAs? > >> >> > > There is no G.723 codec available on Asterisk > >> >> > > I need to know this because of the firewall. > >> >> > > >> >> > if you turn off the reinvite in the asterisk configs for those > >ata186s > >> >> > then it will pass through asterisk even if asterisk doesn't > >understand > >> >> > the codec. > >> >> > > >> >> > -- > >> >> > Steven Critchfield <[EMAIL PROTECTED]> > >> >> > > >> >> > _______________________________________________ > >> >> > Asterisk-Users mailing list > >> >> > [EMAIL PROTECTED] > >> >> > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >> > > > > >> > > > > >> > > > >> > > > >> _______________________________________________ > > > >> Asterisk-Users mailing list > > > >> [EMAIL PROTECTED] > > > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> >-- > >> >Steven Critchfield <[EMAIL PROTECTED]> > >> > > >> >_______________________________________________ > >> >Asterisk-Users mailing list > >> >[EMAIL PROTECTED] > >> >http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > > > [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > > >_______________________________________________ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
