similar to: iax2 group pickup

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:

You have the session target as the IP address of the router's own ethernet interface. You probably want that to be the address of the Asterisk server instead. I also highly recommend you use full duplex ethernet, as voice packets don't really like to be restransmitted when a collision happens. -d > Message: 10 > From: "Bartosz Jozwiak" <bartek@cq-link.sr> >

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

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

Me too, with 3X Snom 360's in CAP positions. I just gave up in the end and cron'd a SIP REBOOT from my * box at strategic times. If your phone supports the REBOOT directive I'd just reboot your phone every day, 4 AM or what have you. -----Original Message----- From: Bartosz Jozwiak [mailto:bartek@cq-link.sr] Sent: Monday, April 24, 2006 12:35 PM To: Asterisk Users Mailing List -

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

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

I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek

Has anyone seen this before, and know what is happening? USER at HOST:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object

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

Hi, With 0.0.6pre3: # ./build.sh CMake Warning (dev) in CMakeLists.txt: No cmake_minimum_required command is present. A line of code such as cmake_minimum_required(VERSION 2.6) should be added at the top of the file. The version specified may be lower if you wish to support older CMake versions for this project. For more information run "cmake --help-policy CMP0000".

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

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

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

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

Dear all, I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and run with chan_capi. Is anybody using that card ? Would appreciate any feedback. Bartosz

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

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):

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