search for: dialnumber

Displaying 16 results from an estimated 16 matches for "dialnumber".

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2009 Dec 23
1
AMI originate and PHP
Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was c...
2015 Apr 27
2
adding area code
here is what I have: exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: > Motty > Yes > From your dial plan accept 9 + 7 digits then conca...
2015 Apr 27
5
adding area code
Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax
2015 Apr 27
0
adding area code
forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. Thanks, On 04/27/2015 02:38 PM, Motty Cruz wrote: > here is what I have: > > exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) > > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) > > exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) > > not having success; > > "Got SIP reponse 503" Service Unavailable" > > On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: >> Motty >> Yes >> Fr...
2015 Apr 27
2
adding area code
...> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) >> >> exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) >> >> not having success; >> >> "Got SIP reponse 503" Service Unavailable? Can you send us the console output when you make the call? -- Cheers, M...
2012 Dec 12
1
Asterisk 11 originate errors
Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1
2015 Apr 27
2
adding area code
On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-7...
2009 Jul 09
1
Dial stops trying after ~30s regardless
Hi, My Dial() is set to the following, but always stops about 30 seconds into the call even when I set it to try for 60 seconds. exten => dialnumber,1,Dial(${DialInfo},60) I am running on 1.6.1-r199820. Is there some other setting that is overriding mine? Or an issue with this release? Thanks for the help. JR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/aste...
2015 Apr 27
0
adding area code
Motty Yes From your dial plan accept 9 + 7 digits then concat your dialed number together with your areacode. This s a brief example. exten => _9XXXXXXX,1,Set(l_HomeAreaCode=555) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; This line should combine your area code and the last 7 digits of your dialed phone number exten => _9XXXXXXX,n,Dial(SIP/${dialnumber},35) Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: &...
2015 Apr 28
0
adding area code
...I am running Asterisk 1.8.22.0 on CentOS. >> >> Thanks, >> >> >> On 04/27/2015 02:38 PM, Motty Cruz wrote: >>> here is what I have: >>> >>> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >>> >>> exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) >>> >>> exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) >>> >>> not having success; >>> >>> "Got SIP reponse 503" Service Unavailable? > > Can you send us the console output when you...
2010 Aug 12
1
Recording the conversation with MixMonitor() ends when the call is transfered
...I notice that when a call that is recorded with MixMonitor is transfered to another co-worker, the recording ends. exten => 409,n,Macro(SDstartrecording,external,${DID}) the incoming call then goes to a queue... [macro-startrecording] ; ARG1 = incoming DID or CALLERID(name) ; ARG2 = outgoing dialnumber ... exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R ${NR}:astDrisk /var/ftp/${NR} && chmod 570 /var/ftp/${NR} && chmod 400 /var/ftp/${NR}/${recordfile}) ... The 'b'-option makes the recording start when the call is answered by an agent of the queue....
2015 Apr 27
0
adding area code
On Mon, 27 Apr 2015 14:30:07 -0700 (PDT) Steve Edwards <asterisk.org at sedwards.com> wrote: > On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > > > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) > > Missing a colon? > > ${EXTEN:-1} > Does that work? I've always seen it like this: ${EXTEN:1} -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0
2014 Jun 29
0
Passing parameters to voiceglue.conf
Hi Freinds, I am trying to do the following. 1. Accept the call from call ifle. 2, Answer it 3. Extract the dial number and variables from the call file request. 4. Pass that parameters to voiceglue 5. Catch the parameters (dialnumber and cli) in voiceglue.conf 6. Point to the voice.xml file dynamically by matching number I want to make the line in voiceglue.conf as <DEST_NUMBER> http://127.0.0.1/vxml/<FILE>.vxml so that I can catch parameters and call the file dynamically. Please advice me. Best Regards, Roy -...
2015 May 04
0
Asterisk proxying a REFER
...t;>> >>> Thanks, >>> >>> >>> On 04/27/2015 02:38 PM, Motty Cruz wrote: >>>> here is what I have: >>>> >>>> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >>>> >>>> exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) >>>> >>>> exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) >>>> >>>> not having success; >>>> >>>> "Got SIP reponse 503" Service Unavailable? >> >> Can you...
2006 May 23
3
AGI ?
Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use