similar to: Tested 7905G

Has anyone ever worked opn TAPI stuff to make asterisk work with it ? I'm a Windoze C++ developer dig'n into asterisk (and linux at the same time) since a few months and i'm quite interested in creating a TAPI driver for asterisk. so if anybody did any research in that way please inform me. Also i've you think it's quite impossible to do it we can discuss our idea's

We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael Sorry for the previous html mail DISCLAIMER: The content of this e-mail message does not constitute a commitment

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

Hi everybody, I will receive my CISCO 7960G tomorrow. I've ordered it as a "global spare" without any callmanager licence. Now I don't know if I can get firmware-updates so could please someone send me the SIP-firmware? Is the default firmware the "skinny" one? Wich would be better to use with asterisk? Thank you very much

Dear All, We are all Windoze C++ developers but are working on C/C++ development for linux. But we need someone of our team to be trained quite soon in Asterisk development to add some features we really need inside the company. We are searching somebody or a company who will train a developer during a week (i estimate) We are from Belgium ... so if there is a experienced Asterisk developer

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

- I'm not a Linux user i'm trying to get in it ... - Fedora core 1 - QuadBRI card bought from junghanns.net - We want to use the card in TE mode to connect to the TELCO - Downloaded BRISTUFF0.0.2(stable) latest from junghanss.net/asterisk - followed the instructions on voip-info.org - compiled everything still got the error when "make load" in qozap : insmod ./qozap.o

Hi I want to connect a Windows machine over TAPI with the Asterisk PBX. So is it possible to connect the Windows machine directly to Asterisk (Zaptel card)? Thanks Harry Baron

Everythings works great with asterisk exept one feature with redirect : it doesn't redirect when ringing ... BTW are their any plans to extend the manager API ?? Michael Devenijn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031212/1200012e/attachment.htm

I have some strange question bout the asterisk (gpl license ...) but i'm not an experienced linux user ... What happens if for example a big company buys digium , do we have a garantuee that asterisk stays opensource ??? Kind regards Michael Devenijn -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3208

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

Hi, You need to have completely replaced the Microsoft driver, because it doesn't support the headset or ctp Bluetooth profiles. This gave me fits! I followed the instructions at http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html and it works with both a Plantronics and a Motorola Headset, and I can answer calls with idefisk, eyebeam, x-lite, and kapanga. If you end

Hello all, I''m mucking around trying to get my desktop computer (Ubuntu 11.10, BlueZ 4.96, ASUS USB-BT211 USB Bluetooth adapter, no microphone hooked up at the moment) to act as a a hands-free device for my phone (Nexus S). It was relatively simple getting it set up for A2DP, and I was able to play music that is on my phone through my desktop''s speakers (using PulseAudio

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

I didn't get any response on that question, so i supose this feature is possible but there isn't an implementation of it. I'm ready to sponsor this feature in the manager interface (i tried the redirect command but it doesn't work) can somebody help me ?? this feature would make it possbile to use drag & drop features ... Kind Regards Michael Devenijn

Hi all, Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP

I am able to pickup, hangup, and flash, using the buttons on the phone with all of the soft clients and both of the headsets I mentioned below. I don't believe I had to change anything on the client side, just had to get the ctp (cordless telephone profile) working in the bluetooth stack, which was a pita. I struggled with the same issue...I could use the headset, but that's not very handy

Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/22a585ea/attachment.htm

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi