similar to: *@home .conf files request

I am trying to compile zapata under a 2.6 kernel (Suse 9.1 all patches installed). I am getting the error bellow: Any ideas? Anybody able to successfully compile this in Suse 9.1? 500@suse91lx:~/dl/pbx/zaptel-1.0-RC2> make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt make -C /usr/src/linux-2.6

> -----Original Message----- > From: Iassen Hristov [mailto:ih.ng@databrokers.net] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens

how can i do to display a message to every wrong number ??? -- Luis Diaz - Un obsesivo con proyectos! :oP

Anyone having problems with registration to * from a SPA-841? I got a SPA-841 a week ago. I noticed that sometime it could not be reached (dialed to) and it can't dial. In this case the line LED is yellow. I enabled logging to syslog and there is a hint as to what happens. For some reason sometimes it gets "401 Unauthorized" Any ideas what is happening and how to fix it? Phone

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

Hi all, im Luis from argentina i started to setup asterisk in my network but i just cant... so i have a few questions, if some one can help me with examples and or explanations would be great! My Setup: Asterisk 1.0.7 (on 10.0.0.254) Gentoo 2005.0 1) Can i have LOCAL users on my lan? 2) what should i use AIX or SIP 3) can my local users have a phone number? (lets say im 1112 and i can call to

I am also very interested in CRM integration. Anything I can do to help? One thing I don't understand is how is the browser being launched on the person's PC. Or is it not launched automatically? Anyone know of a simple app running on the desktop to do this? I looked into IPSwithcBoard and it appears like it should be able to do the job, but: a) It is pretty heavy - it does a lot of

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

(forgive the brief interruption to -users with a mostly -dev issue, just wanted to publicize this on behalf of the larger community) If there are any ambitious coders out there (not too many shekels yet, but I expect some folks may pony-up) please see: www.voip-info.org/?page=Asterisk+Bounty+Email+TTS We are at $150 & counting. Maybe lobby your exec's for $50 to contribute to this,

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

Hello, We use astGUIclient, it does have server side apps that have to be installed on your Asterisk server, but it does have callerID popups that allow you to search a customizable web page when a call comes in. We are also releasing a new version of the astGUIclient app next week that is entirely web-based and easier to configure the client side. http://astguiclient.sf.net/ MATT---

> -----Original Message----- > From: Iassen Hristov [mailto:ih.ng@databrokers.net] > Does this matter? All we are saying is that Exchange supports > IMAP and we > would use IMAP as the protocol to delete the message from the user's > mailbox. How does the user access his mailbox is his choice. I think two threads of discussion got crossed. Somewhere along the line someone

Some calls I get from google voice, I just send myself an email about the call and want to hangup. But I can't seem to make gv know I've hung up. extensions.conf: same => n,GoToIf($["${CALLERID(num)}"="office"]?email) ................. same => n(email),System(/usr/local/bin/emailme........) same => n,Answer() ; also tried without this same =>

Greetings. I am trying to get AMP up and going on my Asterisk server. I can access the admin pages on my asterisk server via a web browser. I can add and edit things via the web browser and it edits the database accordingly. Everything seems fine except when I try to run 'amportal start'. Below is what I get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log

Hello, Great job on the Asterisk@home project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much.

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

Hi!I've installed Asterisk@home and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2) This is how I installed bristuff: how to install hfc card after unload asterisk and amportal whit amportal stop type "setup" unselect zaptel in system service... and set the lan --->reboot<--- cd /usr/src wget