Re: [asterisk-users] No audio format found to offer.
> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Ernie Dunbar > Sent: Wednesday, June 29, 2011 6:34 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] No audio format found to offer. > > Quoting Carlos Chavez : > > > > The disallow line must be set before any allow line. > > > > Since Asterisk has no official G723 support you should > not even be > > trying to use that. > > That's fantastic. I'll tell that to our SIP trunk provider right away. > > > Do you have the G.279 codec and license installed in your system? > > Remember that G.729 is not included in Asterisk (as a > > codec) so it only works in passthru. > > So G.729 will only work for this trunk if the customer's ATA > is using it too? Assuming Asterisk does not have to transcode yes. Transcoding is required to play Asterisk sound files (if g729 versions are not installed), the T/t/W/w option to Dial, ChanSpy, etc. "Pass thru" may sound cool, but it seldom works well in the real world. Spend the money on a G729 license from Digium ($10/channel) and save yourself problems. -- _ -- 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
Re: [asterisk-users] No audio format found to offer.
Quoting Carlos Chavez : The disallow line must be set before any allow line. Since Asterisk has no official G723 support you should not even be trying to use that. That's fantastic. I'll tell that to our SIP trunk provider right away. Do you have the G.279 codec and license installed in your system? Remember that G.729 is not included in Asterisk (as a codec) so it only works in passthru. So G.729 will only work for this trunk if the customer's ATA is using it too? You need to purchase some licenses and install the codec for it to work. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 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
Re: [asterisk-users] No audio format found to offer.
On Wed, 2011-06-29 at 18:12 -0400, Alex Balashov wrote: > 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/t564/1XX4332,,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 1XX4332 > > -- Couldn't call t564/1XX332 > > == 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. > > The disallow line must be set before any allow line. Since Asterisk has no official G723 support you should not even be trying to use that. Do you have the G.279 codec and license installed in your system? Remember that G.729 is not included in Asterisk (as a codec) so it only works in passthru. You need to purchase some licenses and install the codec for it to work. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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
Re: [asterisk-users] No audio format found to offer.
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/t564/1XX4332,,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 1XX4332 -- Couldn't call t564/1XX332 == 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
[asterisk-users] No audio format found to offer.
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/t564/1XX4332,,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 1XX4332 -- Couldn't call t564/1XX332 == 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 data0 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
