similar to: Dell server for asterisk question!

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP:

I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/db22d880/attachment.htm

Dear Users, Recently I have started using HINT option in Asterisk 1.2.4 with my Polycom 500 phone. What I have notice that for a day or two everything is working great but then HINTs stop working on my Polycom phone. It also happens when I reload asterisk from console. I do sip debug and I do not see anymore asterisk sending NOTIFY messages about my watched extension. To make it work again I

Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should "talk" to Asterisk of course.. Thanks very much for some advices :) Bartek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear users: I have bought TDM04B card and it works in PCI 2.2 ver. slot. How can I check if specific mother board support PCI 2.2 ver. I do not have any documentation for that motherboard. B.

Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. Thank you in advance Bartosz

Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex ____________________________________________________________________________________ Do you Yahoo!? Everyone is

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart

Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group.... as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart -------------- next part

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a question. I set up an extension to 1234 exten => 1234,1,Dial(SIP/1234@sip.greentone.com:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

Hi, Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

Hello, I am trying to run festival, it is running but I am getting this when I run tts_ping.agi WARNING[278546]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Enter the eye-p address you wish to ping. WARNING[278546]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... WARNING[278546]: File app_festival.c, Line 400 (festival_exec):

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this

Hi Guys, Sorry about writing to that list but could not find better place. I have Grandstream BT-100 phone, btw, was working great with Asterisk. I have upgraded the phone, and during upgrade something went wrong. Right now when I power the phone I can only see some garbage on the LCD display. does not react on any buttons, pings,..... Maybe somebody has any idea if it is fixable or I can just

In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/6eb392cc/attachment.htm

What is better? Cisco 7960 or Snom 200 ?? Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031028/befeaa4d/attachment.htm

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm