Re: [Asterisk-Users] Problem with Wait() and chan_capi-cm?
Hi! Armin Schindler wrote: there is a bug, I would need a full log (set verbose 5 ; capi debug) to find out. Of course you would, I just didn't know if it was one. But: if there is a call signaled, the switch has a timeout (about 4 or 5 seconds), this timeout can be extended by sending ALERT (Ringing). Okay, is the timeout necessary? Or: is this short timeout necessary? It appears in the sip-context the timeouts are much longer. So you should do exten = 12345,1,Ringing() (or after the unsuccessful Dial()) to make sure the ISDN call gets ALERT in that case. Yes, that does what I expect. My extensions now are: exten = 12345,1,Dial(SIP/me,30) exten = 12345,2,VoiceMail(su12345) exten = 12345,3,Hangup() exten = 12345,102,Set(COUNT=10) exten = 12345,103,While($[ ${COUNT} 0 ]) exten = 12345,104,Set(COUNT=${COUNT}-1) exten = 12345,105,Ringing() exten = 12345,106,Wait(3) exten = 12345,107,EndWhile() exten = 12345,108,VoiceMail(su12345) Could probably be a bit tiedier, but at least it works. Thank you, Florian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Wait() and chan_capi-cm?
Hello Armin, Armin Schindler wrote: On Sun, 12 Feb 2006, Florian Heer wrote: Armin Schindler wrote: But: if there is a call signaled, the switch has a timeout (about 4 or 5 seconds), this timeout can be extended by sending ALERT (Ringing). Okay, is the timeout necessary? Or: is this short timeout necessary? It appears in the sip-context the timeouts are much longer. Maybe, but that timeout is given by your ISDN provider. Ah, okay, I understand, I thought that was a timeout in chan_capi, then I'd have wondered, if there was any reason for that. There is no need to create a loop. Just one Ringing() is enough. The first ALERT (Ringing) signales the switch to incease the timeout from 4-5 seconds to 2 minutes or so. Thank you very much, now I have a working version that also looks nice enough :-) exten = 12345,1,Dial(SIP/me,30) exten = 12345,n,VoiceMail(su12345) exten = 12345,n,Hangup() exten = 12345,102,Ringing() exten = 12345,n,Wait(30) exten = 12345,n,VoiceMail(su12345) Regards, Florian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Wait() and chan_capi-cm?
Hi! I am playing around with Asterisk and have a problem :-) (Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a sip-phone at my desk and an ISDN-phone (independent of the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't answer. The 'normal' extension would be something like this: exten = 12345,1,Dial(SIP/me,30) exten = 12345,2,VoiceMail(su12345) Works fine as long as the sip-phone is available, if it is not, it is flagged congested/busy, so the next extension would be 102, if I wanted VoiceMail to kick in in that case, this works: exten = 12345,1,Dial(SIP/me,30) exten = 12345,2,VoiceMail(su12345) exten = 12345,102,VoiceMail(su12345) But that is not, what I had in mind, I would like to have 30 seconds to get to the phone, so in theory, this should do the trick: exten = 12345,1,Dial(SIP/me,30) exten = 12345,2,VoiceMail(su12345) exten = 12345,102,Wait(30) exten = 12345,103,VoiceMail(su12345) But Asterisk can not take over the line after the wait. To test, if the Wait was the problem, I created this: exten = 12345,1,Wait(10) exten = 12345,2,Answer() exten = 12345,3,Milliwatt() And still: Asterisk can't take over the ISDN line. The console output says: == ISDN1: Incoming call '12345' - '12345' -- Executing Wait(CAPI/ISDN1/12345-19, 10) in new stack -- Executing Answer(CAPI/ISDN1/12345-19, ) in new stack == ISDN1: Answering for 12345 -- Executing Milliwatt(CAPI/ISDN1/12345-19, ) in new stack CAPI INFO 0x34d1: Invalid call reference value == Spawn extension (capi-in, 12345, 3) exited non-zero on 'CAPI/ISDN1/12345-19' == ISDN1: CAPI Hangingup If I try that in a pure sip-context, it works as I thought it would. Now: do I do something wrong? Is there a problem with the Wait() application? Or is that more likely a bug in chan_capi-cm? Regards, Florian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Wait() and chan_capi-cm?
[EMAIL PROTECTED] wrote: Try build 8015. I know its odd, but this is just like the problem I am having... Uhm... sorry if I seem a bit uninformed, but how do I get that version? Regards, Florian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with dialing out and chan_capi
Hi! I have just installed asterisk and got it to work in a way, but I can't dial out via asterisk. Capi seems to be configured correctly as the system can answer incoming calls or route them to a sip phone. But as soon as I try to call out, asterisk just dies. The output in that case is: *CLI -- Executing Dial(SIP/florian-386b, CAPI/1234567:2437826|60) in new stack Killed Ouch ... error while writing audio data: : Broken pipe Warning, flexible rate not heavily tested! Does anybody have a solution? my extensions.conf right now only contains the line exten = _XXX.,1,Dial(CAPI/1234567:${EXTEN},60) the msn is configured correctly, when it wasn't, an error was produced. Regards, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users