search for: _9xxx

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2013 Nov 25
4
Voicemail greeting playback issues?
Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked
2007 Dec 26
2
Two lines for outgoing calls
Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _90900XXXXX,1,Dial(Zap/g1/${EXTEN:1},30m,tTr) When Zap/1 is busy and I try to call, it wi...
2007 Jan 29
3
Pickup() ringing extension and call waiting
...|tTrR) [internal] include => outbound-local include => parkedcalls exten => 200,1,Dial(SIP/somebody1|20|tTrR) exten => 201,1,Dial(SIP/somebody2|20|tTrR) exten => 202,1,Dial(SIP/somebody3|20|tTrR) exten => _8.,1,Pickup(${EXTEN:1}) [outbound-local] ignorepat => 9 exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT) When there is incoming call and extension 200 rings, I press 8200 to pickup a call and I get disconnected. here is debug from asterisk CLI: ? ?-- Starting...
2007 Jun 21
1
TDM400 one way calls
...n => s,102,Voicemail(b${MACRO_EXTEN}) exten => s,103,Hangup ; [macro-oneline1] exten => s,1,Dial(${ARG1},20,t) exten => s,2,Voicemail(u${ARG2}) exten => s,3,Hangup exten => s,102,Voicemail(b${ARG2}) exten => s,103,Hangup ; ; [default] ;setupdial out ; ;test dialplan exten => _9xxx,1,SayDigits(${EXTEN:1}) ; ;setup the phone extensions exten => 400,1,Macro(oneline,${FSEXT1}) exten => 401,1,Macro(oneline,${CUSTSERVE1}) exten => 402,1,Macro(oneline,${CUSTSERVE2}) exten => 410,1,Macro(oneline,${FSEXT3}) exten => 421,1,Macro(oneline,${LONDONSOLE1}) exten => 450,1...
2003 Jul 18
2
Correct syntax to call using IAX and a different UDP port
Hi, Which is the correct syntax to call using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1}) The first one works great, the second one give me this error in the Asterisk console: -- Executing Dial("SIP/351-24f4", "IAX/user:pass@19...
2007 May 22
3
Dial out issues.
...l(b${MACRO_EXTEN}) exten => s,103,Hangup ; [macro-oneline1] exten => s,1,Dial(${ARG1},20,t) exten => s,2,Voicemail(u${ARG2}) exten => s,3,Hangup exten => s,102,Voicemail(b${ARG2}) exten => s,103,Hangup ; [default] ;setupdial out include => from-pstn ; ;test dialplan exten => _9xxx,1,SayDigits(${EXTEN:1}) ; ;setup the phone extensions exten => 400,1,Macro(oneline,${FSEXT1}) exten => 401,1,Macro(oneline,${CUSTSERVE1}) exten => 402,1,Macro(oneline,${CUSTSERVE2}) exten => 410,1,Macro(oneline,${FSEXT3}) exten => 421,1,Macro(oneline,${LONDONSOLE1}) exten => 450,1...
2007 Jul 26
1
tdm400p fxs module busy
...t; s,1,Dial(${ARG1},20,t) exten => s,2,Voicemail(u${ARG2}) exten => s,3,Hangup exten => s,102,Voicemail(b${ARG2}) exten => s,103,Hangup ; [macro-fax] exten => s,1,Dial(${ARG1},20,t) exten => s,3,Hangup ; [default] ;setupdial out include => from-pstn ; ;test dialplan exten => _9xxx,1,SayDigits(${EXTEN:1}) ; ;setup the phone extensions exten => 400,1,Macro(oneline,${FSEXT1}) exten => 401,1,Macro(oneline,${CUSTSERVE1}) exten => 402,1,Macro(oneline,${CUSTSERVE2}) exten => 410,1,Macro(oneline,${FSEXT3}) exten => 421,1,Macro(oneline,${LONDONSOLE1}) exten => 450,1...
2004 Nov 22
0
How to configure the Asterisk server such that a FXS phone can talk to SIP client?
...(usually 1 or 0) ;TRUNK=IAX2/user:pass@provider ; ; test setup!! [pstn-incoming] exten => 999,1,Dial,SIP/999@30.0.0.1; exten => 201,1,Dial(SIP/201,20) exten => 202,1,Dial(SIP/202,20) ;include=> lan-phones ;include=> pstn-outbound include=> autocontext [pstn-outbound] exten=> _9XXX, 1, SIP/${EXTEN}@30.0.0.1; [lan-phones] exten => 201, 1, Dial(SIP/201,20) exten => 202, 1, Dial(SIP/202,20) ; ; Here are the entries you need to participate in the IAXTEL ; call routing system. Most IAXTEL numbers begin with 1-700, but ; there are exceptions. For more information, and to...