Thank you for your help Steven. > Message: 8 > Subject: Re: [Asterisk-Users] Drops due to codecs? > From: Steven Critchfield <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Date: 03 Jul 2003 15:26:45 -0500
> Reply-To: [EMAIL PROTECTED] > On Thu, 2003-07-03 at 15:01, Daniel Flickinger wrote: > > Hello, > > > It is my understanding that on the softphone side, asterisk is only > > responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And > if it does matter, are there any codecs that cause problems with asterisk > bridging two SIP connections? Thanks for your helpful input, > This depends. If a SIP phone must use the IVR feature of asterisk to get > routed to another SIP phone, then codecs matter. If asterisk is > listening on the line to hear when you issue commands via DTMF for it to > do something like transfer, then yes it matters. Also some SIP phones > don't handle reinvite properly, and then asterisk is stuck redirecting > the audio from place to place for you, here it doesn't matter unless on > of the above comments apply. > -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
