Codecs are always an issue. Best to put disallow=all and allow=whatevercodecyouwant in each [sipuser] entry. You can't have Asterisk do codec translation (transcoding) bewteen g729 and some other codec unless you have the g729 licenses (US$10/channel from Digium).
Transcoding would be required for access to ANY of the asterisk sound files, voicemail and PSTN via Zap interfaces. On Tue, 2003-09-09 at 14:51, Ernest W. Lessenger wrote: > At 02:38 PM 9/9/2003 -0500, you wrote: > >That would be reinvite= and canreinvite= in the user entry for each SIP > >endpoint. Asterisk will allow the endpoints to talk directly to each > >other if both those settings are = yes (the default, I think) AND both > >endpoints use the same protocol (SIP) AND the same codec. > > So Asterisk will allow it... and if I set both to no, asterisk would act as > a true proxy, using the most bandwidth efficient codec available for each > leg of the call (i.e. GSM for x-lite and g.729 for Cisco et al)? > > Thanks, > --Ernest > > > >On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote: > > > I have seen this asked in the archives several times, but do not see a > > > definitive answer anywhere. Is there a way to tell the Asterisk to act like > > > a "normal" SIP Proxy, handling only the SIP messages, and letting the > > RTP go > > > point-to-point? > > > ----- Original Message ----- > > > From: "Sean Figgins" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Sent: Tuesday, September 09, 2003 1:40 PM > > > Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet > > > > > > > > > > On Mon, 8 Sep 2003, Jim Mercer wrote: > > > > > > > > > > Can we bribe you? :) > > > > > > > > > > sure, pay my rent for 3 months and give me a 50" plasma TV to play in > > > the > > > > > background. > > > > > > > > Is that all? That sounds rather cheap, compared to the things direction > > > > that I'd have to go if I wanted to stick to the cisci CM route, with > > > > licenses for every endpoint that I want to connect. > > > > > > > > Realistically... I just can not comprehend how to get stuff to work > > > > correctly with Linux. I used to be a Linux nut years ago, but once I > > > > found FreeBSD with it's ports collection, I wondered why anyone ever > > > > bothered with Linux and it's completely messed up software install > > > > requirements. > > > > > > > > Right now, under RedHat 9.0, I have * running, but no hardware, and I > > > > can't figure out how to get h.323 operational so I can talk to my cisco > > > > gateway with the PRI interface... I'm only guessing that FreeBSD > > would be > > > > much easier for non-programmers like myself. > > > > > > > > -Sean > > > > > > > > _______________________________________________ > > > > 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 > >-- > >BTEL Consulting > >850-484-4535 x2111 (Office) > >504-595-3916 x2111 (Experimental) > >877-552-0838 (Backup Phone) > > > >_______________________________________________ > >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 -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
