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
