Re: [asterisk-users] Playback on h exten
2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: -- Executing [301@from-test:1] Dial(SIP/300-0045, SIP/301,60,rjtTg) in new stack -- Called SIP/301 -- SIP/301-0046 is ringing -- SIP/301-0046 answered SIP/300-0045 -- Auto fallthrough, channel 'SIP/300-0045' status is 'ANSWER' -- Executing [h@from-test:1] Goto(SIP/300-0045, play,s,1) in new stack -- Goto (play,s,1) -- Executing [s@play:1] NoOp(SIP/300-0045, play) in new stack -- Executing [s@play:2] SayDigits(SIP/300-0045, 123579) in new stack [Feb 21 10:35:00] WARNING[31945]: file.c:833 ast_readaudio_callback: Failed to write frame -- SIP/300-0045 Playing 'digits/1.ulaw' (language 'en') == Spawn extension (play, s, 2) exited non-zero on 'SIP/300-0045' This is my dialplan: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = h,1,Goto(play,s,1) [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Anyone can help me? Thanks Enrico. If you choose to go with the Dial command and use the g option, you have not to use the h extension, but just provide a next priority. Your dialplan has to be: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) *exten = _X.,2,Goto(play,s,1)* [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Leandro -- _ -- 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] Playback on h exten
Yes, correct now it works for Dial. I think is the same with c option on Queue, do you think there's a way to do it on h exten? My goal is to inject my dialplan on hangup macro. Enrico. - Messaggio originale - If you choose to go with the Dial command and use the g option, you have not to use the h extension, but just provide a next priority. Your dialplan has to be: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = _X.,2,Goto(play,s,1) [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Leandro -- -- Pasqualotto Enrico cell. +39 3473292620 skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto http://www.netspin.it :: e.pasqualo...@netspin.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback on h exten
On Thursday 21 February 2013, Enrico Pasqualotto wrote: Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: . stuff deleted . This is my dialplan: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = h,1,Goto(play,s,1) [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Anyone can help me? By the time you reach extension h, it's too late for playing audio; the call has already been hung up, and the process of freeing up resources has begun. If you are using the g modifier to the Dial() command then the call doesn't jump to extension h, but carries on with the next step if the far end hangs up. So what you probably want is something more like this: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = _X.,2,Noop(play) exten = _X.,3,Saydigits(123579) -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback on h exten
The h exten is triggered when the channel is hangup, so you cannot send any voice data on it. Leandro 2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it Yes, correct now it works for Dial. I think is the same with c option on Queue, do you think there's a way to do it on h exten? My goal is to inject my dialplan on hangup macro. Enrico. -- If you choose to go with the Dial command and use the g option, you have not to use the h extension, but just provide a next priority. Your dialplan has to be: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) *exten = _X.,2,Goto(play,s,1)* [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Leandro -- -- Pasqualotto Enrico cell. +39 3473292620 skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto http://www.netspin.it :: e.pasqualo...@netspin.it -- _ -- 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] Playback on h exten
Any application with audio stream like playback or background will not work in h priority, use of g in Dial() and c in Queue() is best approach for the same. Refer following link for h priority detail explanation http://www.voip-info.org/wiki/view/Asterisk+h+extension Regards, Bharat On Thu, Feb 21, 2013 at 3:53 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Thursday 21 February 2013, Enrico Pasqualotto wrote: Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: . stuff deleted . This is my dialplan: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = h,1,Goto(play,s,1) [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Anyone can help me? By the time you reach extension h, it's too late for playing audio; the call has already been hung up, and the process of freeing up resources has begun. If you are using the g modifier to the Dial() command then the call doesn't jump to extension h, but carries on with the next step if the far end hangs up. So what you probably want is something more like this: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = _X.,2,Noop(play) exten = _X.,3,Saydigits(123579) -- AJS Answers come *after* questions. -- _ -- 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 -- Bharat Lalcheta -- _ -- 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