[asterisk-users] How to pick a codec on the fly
Hi list, I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. Any ideas? Google wasn't really helpful on this one. Danny Nicholas -- _ -- 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] How to pick a codec on the fly
On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba -- _ -- 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] How to pick a codec on the fly
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Monday, September 27, 2010 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pick a codec on the fly On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X. -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, default|s|1) in new stack -- Goto (default,s,1) -- Executing [...@default:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:2] Goto(DAHDI/1-1, select-func|s|1) in new stack -- Goto (select-func,s,1) -- Executing [...@select-func:1] WaitExten(DAHDI/1-1, 5|m) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 == CDR updated on DAHDI/1-1 -- Executing [...@select-func:1] Set(DAHDI/1-1, _SIP_CODEC=ulaw) in new stack -- Executing [...@select-func:2] Dial(DAHDI/1-1, IAX2/xxx/332|30|m) in new stack -- Called xxx/332 -- Started music on hold, class 'default', on DAHDI/1-1 -- Call accepted by XXX.XXX.XX.XX (format gsm) -- Format for call is gsm -- IAX2/ffb-18075 answered DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 -- Hungup 'IAX2/xxx-18075' == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- _ -- 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] How to pick a codec on the fly
i think it's SIP_CODEC now .. and not _SIP_CODEC? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: dan...@tryba.nl; asterisk-users@lists.digium.com Date: Mon, 27 Sep 2010 13:30:08 -0500 Subject: Re: [asterisk-users] How to pick a codec on the fly -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Monday, September 27, 2010 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pick a codec on the fly On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X. -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, default|s|1) in new stack -- Goto (default,s,1) -- Executing [...@default:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:2] Goto(DAHDI/1-1, select-func|s|1) in new stack -- Goto (select-func,s,1) -- Executing [...@select-func:1] WaitExten(DAHDI/1-1, 5|m) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 == CDR updated on DAHDI/1-1 -- Executing [...@select-func:1] Set(DAHDI/1-1, _SIP_CODEC=ulaw) in new stack -- Executing [...@select-func:2] Dial(DAHDI/1-1, IAX2/xxx/332|30|m) in new stack -- Called xxx/332 -- Started music on hold, class 'default', on DAHDI/1-1 -- Call accepted by XXX.XXX.XX.XX (format gsm) -- Format for call is gsm -- IAX2/ffb-18075 answered DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 -- Hungup 'IAX2/xxx-18075' == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- _ -- 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] How to pick a codec on the fly
From: da...@debsinc.com To: dan...@tryba.nl; asterisk-users@lists.digium.com Date: Mon, 27 Sep 2010 13:30:08 -0500 Subject: Re: [asterisk-users] How to pick a codec on the fly -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Monday, September 27, 2010 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pick a codec on the fly On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X. -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, default|s|1) in new stack -- Goto (default,s,1) -- Executing [...@default:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:2] Goto(DAHDI/1-1, select-func|s|1) in new stack -- Goto (select-func,s,1) -- Executing [...@select-func:1] WaitExten(DAHDI/1-1, 5|m) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 == CDR updated on DAHDI/1-1 -- Executing [...@select-func:1] Set(DAHDI/1-1, _SIP_CODEC=ulaw) in new stack -- Executing [...@select-func:2] Dial(DAHDI/1-1, IAX2/xxx/332|30|m) in new stack -- Called xxx/332 -- Started music on hold, class 'default', on DAHDI/1-1 -- Call accepted by XXX.XXX.XX.XX (format gsm) -- Format for call is gsm -- IAX2/ffb-18075 answered DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 -- Hungup 'IAX2/xxx-18075' == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 27, 2010 1:40 PM To: Asterisk Users Subject: Re: [asterisk-users] How to pick a codec on the fly I think it's SIP_CODEC now .. and not _SIP_CODEC? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 FWIW, SIP_CODEC is value for use in Asterisk 1, _SIP_CODEC passes the value on to Asterisk 2. -- _ -- 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] How to pick a codec on the fly
From: da...@debsinc.com To: dan...@tryba.nl; asterisk-users@lists.digium.com Date: Mon, 27 Sep 2010 13:30:08 -0500 Subject: Re: [asterisk-users] How to pick a codec on the fly -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Monday, September 27, 2010 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pick a codec on the fly On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X. -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, default|s|1) in new stack -- Goto (default,s,1) -- Executing [...@default:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:2] Goto(DAHDI/1-1, select-func|s|1) in new stack -- Goto (select-func,s,1) -- Executing [...@select-func:1] WaitExten(DAHDI/1-1, 5|m) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 == CDR updated on DAHDI/1-1 -- Executing [...@select-func:1] Set(DAHDI/1-1, _SIP_CODEC=ulaw) in new stack -- Executing [...@select-func:2] Dial(DAHDI/1-1, IAX2/xxx/332|30|m) in new stack -- Called xxx/332 -- Started music on hold, class 'default', on DAHDI/1-1 -- Call accepted by XXX.XXX.XX.XX (format gsm) -- Format for call is gsm -- IAX2/ffb-18075 answered DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 -- Hungup 'IAX2/xxx-18075' == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 27, 2010 1:40 PM To: Asterisk Users Subject: Re: [asterisk-users] How to pick a codec on the fly i think it's SIP_CODEC now .. and not _SIP_CODEC? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Good and Bad news On a SIP call, SIP_CODEC still works; this same patch not built into iax (apparently we don't want Asterisk 1 to set the codec for Asterisk 2). -- _ -- 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
