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...it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned regarding having an open channel between the two machines and there is scant information about setting up a more secure connection. Can anyone offer any good links or howtos for this? The CRM is vtiger and I couldn't see any references to ssl in the...

<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 the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting caller-id on the phone's display, so I guess that shows FSK works from the card to the phone. I did some searching before posting, found http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20zapata.conf which says "On supported hardware, the message waiting light <http://www.voip-info.org/wiki-PBX+Message+Waiting+Indicator> will also be activated ? this probably requires that you also set adsi=yes. Update: This option does NOT require ADSI. It will send a standard FSK...

Hello, i got a question, i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are

On request, I've updated the following page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20indications.conf with ring signals from Brazil. (And at the same time, the Brazilian signals was added to the CVS). If you have an entry in indications.conf that you want to share, a variant or configuration for a country not defined in indications.conf in the distribution - please add it to the...

Hello! I have been googling a lot and asked wiki a few times now, but i cant find a howto for setting up a voicebox. Any link/hint would be great! Thanks, Mario

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

I have a * serving 15 sip clients. I use the digium 4 port t1 card. We have an autodialer that calls and reminds clients of there appointment. it uses a pri t1. I would like to plug its t1 output into asterisk to use voip. I am very new to * and am confused. Any help would be appreciated. _________________________________________________________________ Express yourself instantly with

How do I test if the card is working or not ? Is there something that I can do to get a response from the card ? Ive put the card in, installed drivers ect but can't dial out and can't see a response when I try dial in from external number. Any ideas ? Thanks Shaun --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.857 /

Thanks to Pau (the original person to pose the question on this list), it's fixed. The firewall was getting in the way. I needed to open up UDP ports 10000 to 20000 for RTP traffic. See the following for more info: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config% 20rtp.conf http://www.voip-info.org/wiki-Asterisk+firewall+rules -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/0dce1168/attachment.htm

Hi list, I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't

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

...to read in a log file, a (reliable) copy of AMI originate requests. Any suggestion ? If this helps, I can use astmanproxy. I also saw a debug=on option inside /etc/asterisk/manager.conf but I couldn't find a change in produced output. And http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.confis not helpful on this specific option. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100713/c0679d19/attachment.htm

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. --------------

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

Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search & read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to,

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

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:

You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume > -----Original Message----- > From: Stefan Gofferje [mailto:stefan@gofferje.homelinux.org] > Sent: Friday, February 04, 2005 1:43 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject...