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?
Then the call is passed through Asterisk without any conversion? How can I do to pass all the calls through Asterisk, even if a codec conversion is required or not? Thanks, Dan ----- 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
