similar to: WAMi - Windows Asterisk Manager

I'm writing a program to quickly generate SIP<Mac>.cnf files for tftp configuration of the 79XX series of Cisco phones and would ask that anyone who is interested in using this please send me working examples of your SIP<Mac/Default>.cnf, and RINGLIST.dat files. Also, please send me your ring tones and * related logos for possible inclusion. Thanks, Christian Hoffmeyer YottaDot

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

> I've personally put up a $300 USD bounty on a win32 call manager - > hopefully a few others will help get the ball rolling : > http://bugs.digium.com/bug_view_page.php?bug_id=0000848 Is C# and .NET fine? This is already nearly done. I can send you binaries of a single user call manager, and the operator manager is in the pipe. Actually, I'll just post these for download.

I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 >>>>> > Connected to localhost > Escape character is '^]'. > Asterisk Call Manager/1.0 > Then I type > Action:Login (enter) > Username:sam > Secret:sam > Then I enter twice > > I get Response: error

I am trying to send calls from an AS5300 to Asterisk via e1 and I get this bit of information in place of routing information Going to extension s|1 because of Complete received Accepting call from '' to 's' on channel 1, span 1 Here are the relevant zaptel and zapata pieces. span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 signalling=pri_cpe switchtype=national

Hello all, My set up is a fractionated t1, with 1-6 voice channels, 21-24 data channels. I have a t100P installed in a amd 1500 with 512meg. Question: In order to set up the data channels, is the following correct? 1) compile new kernel with generic hdlc support 2) compile hdlc from hq.pm.waw.pl/hdlc 3) uncomment KFLAGS+=DConfig_ZAPATA_NET in zaptel makefile 4) compile zaptel, 5 compile rest

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

HI I want to know Why large enterprises (F500) are not shifting to asterisk as it is going to save them a lot of investment. Are there some problems with asterisk ??? Varun Gupta India __________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail

All, I am trying to configure hdlc support on T100P Digium card - everything seems ok... but it just does not work at all. I was able to compile all drivers, the light on the card becomes green when I plug T1 link.... but I even can't ping default router IP... there is no data coming back to me, so I am confused completely now. Also I tried both protocols - raw hdlc and cisco hdlc. The

Hello * community, I have 2 * boxes spanning a t1 with hdlc. I'm trying to route packets so people on a subnet seperate of the * boxes can browse the boxes on the * spanned subnet. The * boxes and the boxes on subnet controlled by * can see everything. The boxes on the seperate subnet inside the company can only see so far as the first * box and can't ping across the span.

I know an email can be sent when a user get a voicemail message, but is there a way to send a message to a SIP phone to say they have a message? Or how hard would it be to write an app that could popup on a PC when there is a message in the mail box? Kyle

Hi, I downloaded Asterisk using this command a couple of weeks ago... # cd /usr/src # export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot # cvs login # cvs checkout zaptel libpri asterisk Can someone tell me what I need to type in order to get the latest stabe version when I rebuild my server ? Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed...

Is there someone on the list who has successfully installed one of the packages that integrates Asterisk with Outlook? I've tried but been unsuccesful thus far. I'm looking for guidance on which works well. I'm using Outlook 2003 on WinXP. Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

Hi, This is our test setup: 4 phones, 2 logged into the one queue, the other 2 phones are used to dial into the queue. If there are 2 calls coming into the queue at the same time, we would like to have the 2 queue phones ringing at the same time (one for each call). But as it is, the 2nd phone only starts ringing after the first one was picked up. Scenario: One of the non-queue phones dials

I am terribly sorry to bother the list with such generic and bizarre problems, but I have been racking my brain with these for the last week working on it for at least 60 hours. If anyone can even point me in the right direction I would be eternally grateful. So without further adu here are my woes: I have * (2004-04-09 CVS) running on a P4 1.6Ghz CPU, 512MB RAM, Debian "Sarge", and

Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From: