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
