Perhaps do this instead? allow=g723 allow=g729 disallow=all
On 06/29/2011 05:57 PM, Ernie Dunbar wrote:
This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error: -- AGI Script Executing Application: (DIAL) Options: (SIP/t564/1XXXXXX4332,,HR) == Using SIP RTP CoS mark 5 [Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio format found to offer. Cancelling call to 1XXXXXX4332 -- Couldn't call t564/1XXXXXX332 == Everyone is busy/congested at this time (0:0/0/0) I've checked to ensure that both formats are loaded into Asterisk: voip2*CLI> module show like 729 Module Description Use Count format_g729.so Raw G729 data 0 1 modules loaded voip2*CLI> module show like 723 Module Description Use Count format_g723.so G.723.1 Simple Timestamp File Format 0 1 modules loaded So I'm at a bit of a loss as to why Asterisk is complaining that there's no audio format found to offer. ---------------------------------------------------------------- This message was sent using Lightspeed.ca's Advanced Webmail. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
