similar to: Analogue Line to Asterisk (Which Digium Model???)

Hi, Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS CVS Head I am a newbie and about to install Asterisk on SUSE Server. Can someone please advise what is the best version type and number should I use. My environment is not so big. I only wish to eventually get my asterisk to talk to Cisco CCM 3.3.4.

Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid -------------- next part -------------- An HTML attachment

Dear list, I am running Asterisk CVS-v1-0-12 what is this called in terms of Asterisk versions convention? Is it Stable , Head, latest release !!! Excuse me if the question is too basic, but your help is appreciated. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have => 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXXXXXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XXXXXX instead of

Hi, I just want to know what is the easiest way to have Asterisk route calls to PSTN. Hope any one can help me. PS: Any solution using a Cisco device is preferable. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050112/1604835f/attachment.htm

Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050116/f75bd2fe/attachment.htm

I'm setting up a small office with about 8 SIP phones. Incoming and outgoing lines will be through IAX. We would also like to use an analog line for 911. Is the TDM01B a good option for this kind of configuration? Are there gotchas I'm missing? Finally, we would like to be able to use analog fax machines in the office. Would it make more sense to purchase the TDM400 card with 1 FXO

Hi All, We have just installed Asterisk@Home. It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated.

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

I have installed Asterisk@Home on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2743 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060615/711bfb60/attachment.jpeg

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

Hi All, I have posted this messgage yesterday in the Beowulf mailing list, and did not get any responses, as i have tried different Centos kernels to see if the behaviour changes or not, and it did not change much, I am posting it here, I hope no one minds, and thanks in advance for any pointers or clues? ---------- Forwarded message ---------- From: Walid <walid.shaari at gmail.com>