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...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 Michael Devenijn DKMA bvba -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031112/1df3eb01/attachment.htm

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

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

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

...a SIP image (latest download) and it works great ! for a reasonable price but with a good quality and a brand ... which inspires trust and helps selling better The only minus : Missing a microphone to work handsfree (or i didn't find it.) but strange enough their is a speaker ... Michael Devenijn IT DKMA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031020/0e993890/attachment.htm

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 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040102/9cb61894/attachment.bin

...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 ________________________________ Van: Michael Devenijn Verzonden: ma 24/11/2003 14:38 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] SIP to SIP redirect while ringing is it possible to transfer a call while it's ringing ?? SIP/cs001 calls SIP/cs002 The SIP/cs...

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

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

Hello, I've been lurking around the mailing list and browsing around on Asterisk-related links while I wait for my X100P to come in the mail. So far I haven't found very much information related to what I want to do with Asterisk. I was wondering if someone could point me in the direction of any work that may already have been done on a project similar to the one I'm trying to

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

...son who knows the Asterisk sources very well ! Please can somebody make use a good offer for a training of 1 week in Asterisk development, it is quite logic that this will result in Open source contributions ... and wil bring some extra commercial experience to Asterisk. kind regards Michael Devenijn DKMA bvba Schaarbeeklei 636 1800 Vilvoorde Tel.: +32 2 255 10 19 Fax : +32 2 251 03 12 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031126/9b71af10/attachment.htm

Is it possible to modify the caller id on the phone during a call (session) ? If not does anybody know with which SIP request this could be handled ? I'm know investigating RFC3311 which seems to offer an answer but if somebody already has an answer ... Michael

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

Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume)

- 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 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'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark