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Hello everybody. I am trying to configure an Asterisk on Debian with the Billion ISDN card. I am using mISDN. But when I call on the CLI apears this: -- Executing Dial("SIP/101-081805b8", "mISDN/1/943833473|45|tTwW") in new stack -- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] --> we have already send Release_complete == Everyone is

...exten => 102,2,Hangup exten => 103,1,Dial(SIP/103,30,Ttm) exten => 103,2,Hangup exten => 104,1,Dial(SIP/104,30,Ttm) exten => 104,2,Hangup include => outgoing [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =>_9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() [default] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Dial(SIP/101,30,Ttm) Why is that? Thanks everybody. Have a nice day!!! -------------- next part -------------- An HTML...

...the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten =>_9XXXXXXXX,1,Dial(mISDN/1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() [default] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Dial(SIP/101,30,Ttm) misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/m...

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls are routed to the PSTN perfectly. What is the correct way to route some calls from Asterisk to another h...

...s ;clearglobalvars=no ;priorityjumping=no [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =>_9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() Command line: modprobe zaptel modprobe wcfxo modprobe wctdm Then I start Asterisk (asterisk -vvvc), and when I call to the analog line number, the console shows that: *CLI> -- S...

...gt; >Content-Type: text/plain; charset="us-ascii" > > >Here is my configuration for MEdiatrix 1204, by default the 1204 strips one >digit, so it is not necessary to use: > >To dial OUTSIDE > >EXTENSIONS.CONF > >[locales] >;ignorepat => 9 exten => _9XXXXXXXX,1,Dial(SIP/${EXTEN-1}@Mediatrix) exten => _9XXXXXXXX,2,Congestion exten => _9XXXXXXXX,102,Congestion To receive calls [from-pstn] ;Incoming calls from Mediatrix 1204, the 1204, sends an invite to 1111@110.10.200.2 exten => 1111,1,Dial(SIP/100,20) exten => 1111,2,Voicemail(u100) exte...

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.... > > > > My question is: can Asterisk route some calls thru a second h323 gateway (a > > h323 <-> PSTN gw)? > > > > - Asterisk ip: 192.168.1.10 > > - h323<->PSTN gw: 192.168.1.20 > > > > I've tried: > > > > exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) > > > > or > > > > exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) > > I guess that "192.1.1.20" is a typo, right? > You will have to give more info in order to be able to > find the problem. > Try to set these pa...

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

...NSOLE\})\par ;exten => 8500,1,VoicemailMain\par ;exten => 8500,2,Goto(s,6)\par \par \par [default]\par exten => 1000,1,Dial,Zap/1,20\par exten => 1000,2,Voicemail,u1000\par exten => 1000,3,Hangup\par exten => 1000,102,Voicemail,b1000\par exten => 1000,103,Hangup\par exten => _9XXXXXXXX,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => _9XXXXXXXX,2,Congestion\par \par exten => _9XXXXXXXXXX,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => _9XXXXXXXXXX,2,Congestion\par ; Extension 2000 Sipura line 1\par exten => 2000,1,Dial,sip/spa2000|30|t\par exten => 2000,2,Voicemail,u2000\par ;Exten...