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Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

<http://voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf%20sorting> This page on voip-info.org describes how it is possible to affect the sort order of patterns in extensions.conf. What is doesn't explain is how asterisk really does sort patterns. How does this happen? Adi

I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco

Hey all, Someone's posted one of my 800#'s on a poster in California for free concert tickets, so I'm getting calls from California AC's at all times of the day asking for tickets. I'm just using the 800# for friends and family, and don't know anyone in these area codes, so I'd like to just give these callers either congestion or a prerecorded message. Works fine

This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a

Hello, i hev a subscription to a international voip provider and I want all calls for numbers _001xxxxxxxxxx to go through my voip provider. I tried many settings in sip.conf , extensions.conf and iax.conf. Please give me some simple example for how can i transfer the specified calls to my external voip provider. What may I put and where in witch file. Thank you for your support. --------------

On Tue, Jun 30, 2015 at 12:30:35PM -0700, Eric Lindblad wrote: > I don't know if it would be of any use - and I know that > the configure script searches for gawk, sed and grep - > but Watcom has free 32 and 64 bit compilers, and > the msvc files for Xapian no longer appear to be > maintained, so perhaps Xapian build files written for > [Open] Watcom could be made, as such

Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn! I have several users configured (101, 102, 105, 155, 211, etc). They are all in different

Is there a way to match the last 7 digits of an extension? So that 1008014445454 8014445454 4445454 Would all match? I have looked at extension matching and I can't figure out how to do this:-) Thanks in advance! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Does Asterisk, provide the option within its voice mail system to be able to do the following: We want to have an option to be able to have two mail boxes on one ext. For example: we want to be able to have a French Mailbox and a English one... So I can say the following... You have reached ext 000 to continue this message in French press example 3 - and it would then go to my French