[asterisk-users] Forcing a CODEC
Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- 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] Forcing a CODEC
That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- 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 -- _ -- 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] Forcing a CODEC
Remove all other codec On Nov 15, 2011 8:17 PM, Jaap Winius jwin...@umrk.nl wrote: Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Forcing a CODEC
The variable for outbound is (SIP_CODEC_OUTBOUND=g722) But I think asterisk will try to transcode then because the preferred codec on the phone is ulaw or so -Original Message- From: Danny Nicholas da...@debsinc.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 15 Nov 2011 08:50:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Forcing a CODEC That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- 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 -- _ -- 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 -- _ -- 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] Forcing a CODEC
Hi Thats is also one of the reason On Tue, Nov 15, 2011 at 20:27, isr...@gmail.com wrote: The variable for outbound is (SIP_CODEC_OUTBOUND=g722) But I think asterisk will try to transcode then because the preferred codec on the phone is ulaw or so -Original Message- From: Danny Nicholas da...@debsinc.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 15 Nov 2011 08:50:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Forcing a CODEC That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- 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 -- _ -- 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 -- _ -- 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 -- Amit Anand +91 9818559898 -- _ -- 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
