similar to: want to set a var in sip.conf

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> 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 =>

This was posted at The Asterisk Blog Forums <http://asteriskblog.com/forum/> Click here for the original post.<http://asteriskblog.com/forum/viewtopic.php?t=20> I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore

On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz <motty.cruz at gmail.com> wrote: > here is what I did and worked for me: > > exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) > > exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have a leading underscore on your patterns. Your users

Hello All, Hope you all are fine and good ... I m facing an odd problem. I am able to dial some local and Mobile Number and some are not working. I also try to remove and re create dial plans , but it's not working. I live in New Delhi and Area Code is {011} but i am bale to call the number without adding {0} by just {11}, Previously i was thinking it was PRI problem, so i called the

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

i am using the freebsd port, which seems to not yet have WaitExten(), which i kinda want to use thusly [ext-666] exten => _.,1,SetVar(areacode=666) exten => _.,2,Background(zz-in-who) ; give them list of extns exten => _.,3,WaitExten(10) ; let them enter extn to call include => extensions include => applications include => speeddials

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I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid="crisco" <142>

I've installed asterisk@home and gotten inbound calls going to an extension, extension to extension calling works but I'm still missing a few pieces. The most annoying one is that apparently asterisk is stripping the area code from the number I'm dialing but I can't figure out how to stop it. I have in my outbound route under "Dialing rules": 1NXXNXXXXXX NXXNXXXXXX We

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

Is it at all possible to set a Global Variable freely whenever a context gets used without having to enter an extension priority to use SetGlobalVar? This is really limiting the dialplan for me. Heres an example of what I would like to be able to do. [globals] AREACODE= [local] exten=_NXXXXXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah) [anyoldcontext1] AREACODE=313 include=local [anyoldcontext2]

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to be hands on for each new phone number deployed... so I would like to set up some administrative extensions that can record greetings... lets say: [admin] exten => 8(NXXNXXXXXX),1,Record($1|-greeting.gsm) [incoming] exten => _(NXXNXXXXXX),1,Playback($1|-greeting) exten => _(NXXNXXXXXX),2,Goto($1,1000) exten

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

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was