search for: nxxnxxxxxx

...eive 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=${EXTEN}) exten => _*NXXNXXXXXX*NXXNXXXXXX*,2,Set(__ANI=${CUT(ANI,*,2)}) exten => _*NXXNXXXXXX*NXXNXXXXXX*,3,Set(DNIS=${EXTEN}) exten => _*NXXNXXXXXX*NXXNXXXXXX*,4,Set(__DNIS=${CUT(EXTEN,*,3)}) exten => _*NXXNXXXXXX*NXXNXXXXXX*,5,SetVar(__TransferToExt=6101) exten...

Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com

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 => _(NXXNXXXXXX),102,Playback(generic-greeting) [2122222222] exten => 1000,VoiceMail(22222) [3103333333] exten => 1000,VoiceMail(33333) The concept...

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

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

Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]----sip----[asterisk2]----PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo

Hello, How would I setup where I call into my number and press say 911 and then it would ask for a pass and would accept it and then would prompt for a number so I could call out of my number on the road? Joshua __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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 aren't literally dialing the N's and X's are they? Second, what's with the plus in the extension? You want your users to...

I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)?

...nsion, 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 are required to dial all 10 numbers since there are 3 area codes in Atlanta now. Using freePBX admin. I think asterisk@home is version 2.7. One less than the most recent, in any case. Any suggestions would be helpful. Thanks, Jim. -------------- next part -------------- An HTML...

Is it possible to strip the plus sign from the ${EXTEN} var on an incoming call? We have our system setup to deal with incoming calls to numbers without a plus sign, lots of AGIs and databases we don't want to have to change. We have seen things like this ${EXTEN:1} which you can use in the dial command but we want to basically change the ${EXTEN} var right off when it comes into

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...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 normal NXXNXXXXXX format before I process through all of my dialplan rules? Effectively it means being able to alter ${EXTEN}. Is this doable in any way? b.

...if that offsite location is still on a call forwarded to it by Asterisk. This way a receptionist could choose to transfer calls to a mobile phone only if it's finished with the last call the receptionist forwarded to it. If I configure a custom extension with the destination SIP/TrunkName/NXXNXXXXXX, the calls transfer fine but don't show as busy using the Flash Operator Panel (as an example). Any thoughts? Thanks in advance, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium....

...al by mine TRIXBOX, dial successfully from there PRI PHONE. So may be there is some thing in trixbox. According to me {0} should not dial automatically, so is there is any clue ??? how to fix that,?? Mine Trunk Rule is {11+NXXXXXXX} Mine Outbount Route is : 8| 80|0. 80|XXXXXXXXXX 895|XXXXXXXXXX 8|NXXNXXXXXX 8|NXXXXXX 8|XXXXXXXX 8|[9]XXXXXXXXX I used 8 for dial outbound.... I am using Trixbox.... Thanks in advance -- Thanks Aman Dhally -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090430/bc2dc8d2/atta...

...er etc work fine. When i try to make an external call i am getting message "All routes are busy". In the asterisk console i am seeing "Everyone is busy/congested at this time". In AMP - Outbound dialing i hv configured a route which i call 'local'. The dial pattern is 1NXXNXXXXXX NXXNXXXXXX NXXXXXX and using trunk ZAP/g1. Any idea why i am unable to make outbound call. thanks for your help. -B __________________________________ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

> 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 =>

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

i have extensions in locations across a number of telco area codes. when someone in seattle picks up and dials 91234567, it would be nice to transform it to 92061234567. i would prefer not to have an extension context per area code. it would be cool to be able to set a variable in the sip.conf bit for each phone with it's geographic default area code. or other folk may have a better hack.