similar to: Call logging

Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/b80fa496/attachment.htm

Hello All, Does anyone know of a software emulator that can be used to simulate hardware such as an E1? I need to play with AstUnicall in a test environment and don't have access to these circuits from the US. If there is an alternate way to test/play with AstUnicall, please let me know! Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed...

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr

I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Par?ina Lama Computers Split

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? Another question, when I recive a phone call, can that

Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since.... -- Chris ----- Original Message ----- From: "Chris Earle

Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

I have posted this to the Asterisk Forums, but got no response yet. Sorry if you are reading this for the second time. What fax hardware do I need for a T1? Ideally, I will switch my T1 to a digital PRI (not CAS I'm told, which is not as good) coming into the building. My CLEC said I can do this switch no problem. I have an analog T1 coming in now. >From the Asterisk box, I will connect

For anyone interested, our company released a PHP/MySQL based content manager for the Cisco 79XX series IP Phones compatible with the SIP load yesterday. It's available via: http://www.sourceforge.net/projects/open79xxdir Best wishes, -Corey ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

HI, can somebody tell me how and where must I put the SIP register line? I think is in [general] section of the sip.conf and that I have to put: register => user:password@host:port/localextension but, user and password of the SIP gateway? Because I'm trying this and doesn't work... thanks a lot in advanced michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens