similar to: virtual extension per user ?

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 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 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 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 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

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.

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

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

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

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

Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My

When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear "transfer". I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear

Does anyone know how to setup a linear type of queue strategy? By that I mean that agents will be tried in a particular order and the call will be routed to them unless they are on the phone or not logged in. I want a 3rd party app to be able to re-arrange this order on the fly based on sales and other metrics. Anybody setup something similar? Any pointers or products already out there open

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

Hi. I'm having trouble controlling the user info when sending e-mails from asterisk via sendmail to a Microsoft exchange server. When I receive the email the sender is always "asterisk@TheDomainISpecify.com" and the name of the sender is always "Added by portage for asterisk". I want to change both sender-address and the name of the sender. I'm using Gentoo for my

First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will

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

- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is <line button="4"> <featureID>9</featureID> ... For speeddial is <line button="5"> <featureID>2</featureID> <featureLabel>341</featureLabel> <speedDialNumber>341</speedDialNumber> </line>

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