similar to: which OS would be fine for asterisk

Hi list, Anyone knows where I can get information about configuring a Dialogic card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody told me that I had to buy the driver, but I don't know if this is true and if so, who, how and how much...

Can anyone that can & is willing to do some IAX usability testing from Costa Rica, San Jose to my Asterisk PBX please pmail your contact info thx tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080210/b0b0d6e3/attachment.htm

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some "backtracing".) The ones I know are: -1: no hint for the extension 0: registered && idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere.

Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Awhile back I had heard some talk, in this list I believe that Agent callback login was going to be deprecated in 1.4, I see it is still there. Does anyone know what is happening with this? -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 voip at rockynet.com

Has anyone seen this before? These phones are behind an edgewater. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bKaf87f1c9f;received=xxx.xxx.xxx.xxx. From: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=23943befc9dc103 To: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=as2c0b7dcd Call-ID: 723559d029d27c820c8dae4b01e45c77@192.168.50.31 This phone is

how can I fix this just started ...... Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid

If it is posible for a imcoming call to ring both the Polycom desk phone and my cell phone at the same time, if I dont answer fall back to my voice mail box. I would like to hire someone to cofigure that for me. Bob -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains -------------- next part -------------- An HTML attachment

Hi all! I have an issue with TDM400P FXO card. When a call enter into my IVR and select the proper option, the person that ansswer the call say your "thanks for contact us ..." but the caller cant hear this words because a delay between asterisk and caller part or between asterisk and the ATA device. What is the item on zapata.conf that can affect this delays. Thanks for any help

Would like wish to ALL a Marry Christmas and a happy new year, full of peace, love, happinesses and much success. That let us have one excellent year of 2008. Best Regards Josue Conti.

Hello, A user who uses my Asterisk made me part of a worry about listening to his voicemails. He has received 4 voicemails on January 3, respectively at 3H00 pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received notifications by e-mail at these times. On first listen to his messages, at 8.00 pm, Asterisk has announced two new voicemails(15H00 and 15H36). He has erased thos voicemails. At 8.30pm ,

Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them "production stable" since half the time a new one comes out, a fix for it comes out the next day. So... that said, what's a good version to linger on? I don't *need*

hi, i'm new to llvm. i'm kind of confused as to what llvm is. i think it said somewhere on the site that it's not a language, it's just used for creating languages. but the people at c-- point me here. so i just want to code in assembler with perhaps some higher-level constructs. will llvm let me do this? also, does llvm support simd up to sse4? and does it have a framework

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

Guys, I have a Packer build procedure, that works like a charm, when with CentOS 6.7. Exemplifying: 1- Packer (with virtio disk) + QEmu + CentOS 6.7 ISO; 2- Create a RAW image; 3- Convert the RAW image into QCOW2 for KVM hypervisors (okay); 4- Convert the RAW image into VMDK for ESXi hypervisors (okay). However, when doing the very same procedure, with CentOS 7.2 ISO, it does not boot

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the

Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful.

Hello, I followed the steps by Russell* http://www.venturevoip.com/news.php?rssid=1980* and I got it working for publish_event only. As soon as I add subscribe_event, Asterisk doesn't start and I just get the following message: *Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node ffffffff* I have no idea what's wrong. There is not very much information about this issue. Where