similar to: string length in dialplan

For our internal fax machines, I'm checking if the faxes are going to branch offices. If they are, I want to capture and email them to the branches. I've set up extension 8447 to test this. A fax machines is connected via an SPA 2102 on 173. Any calls from 173 are sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten

Hello, I'm trying to implement multi-party calls according to these instructions: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO They are almost working, except that the Goto at the end of [dynamic-nway-start] doesn't seem to work. When I turn verbosity up a bit, I get something like this in my error log: == Channel 'SIP/SWG-0085a180' jumping out of macro

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub

Hi I have a 3 level nested while-endwhile loop in a macro that when the execution reaches endwhile, it is jumping out to the While at the caller macro. It shouldn't since the are instructions after the endwhile. -- Executing [s at macro-call-from-outside:72] EndWhile("DAHDI/i1/1234567-4a7f", "") in new stack == Channel 'DAHDI/i1/1234567-4a7f' jumping out of

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

Hello I haven't found an example on how to compare the value of a string variable with spaces in it, and the While loop below never exits: ========== extensions.conf exten => start,n,Set(MYVAR="Dummy value") exten => start,n,NoOp(${MYVAR}) ;BAD TOO ;exten => start,n,While(!$[${MYVAR} : "Some string"]) exten => start,n,While($[${MYVAR} != "Some

Hello, I am looking for a loop unfolding procedure implemented in LLVM that helps to transform a while-loop to n-layer If-statements. The transformation should be on IR, although the example below is illustrated on the source level. original loop: * WHILE (condition) DO action ENDWHILE* Expected unfolded loop (2-layer): * IF (condition) THEN* * action* * IF

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Johnathan Falk Network Administrator Clinton Community Schools

I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the

I seem to be having a conceptual problem with the "While" and "EndWhile" applications. It seems that on the first cycle, even if the result of the "While" is false that the enclosed applications will get run. Is this expected? It seems to be counter-intuitive, but I don't know what the intent of the While routines is. I could of course put a

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: 1. A SIP friend dials 102 extension. 2. Asterisk PBX responds with some beeps. 3. The sip friend hangs up the phone. 4. Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. I tried to implement this using h extension but I got the following

I hope this isn't a FAQ. Per the man page I see ways to control the blocksize for hash comparison reasons, but no way to control it for i/o performance reasons. I'm using rsync to copy folder trees full of large files and I'd like to have control of how much data is read / written at a time. Maybe read 10 MB, write 10 MB, etc. Is there an existing way to do that? == details ==

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the