similar to: OT: Open source CRM systems with * integration

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/e71c34a3/attachment.htm

hello everyone, I just had a thought on this subject why not create a daemon process on the Client PC That registers its self and What phone the user is connected. An AGI script could monitor the progress and when answered could send a push to the registered daemon which would push a link to the registered daemon on the telephone operators on the desk top. this would not waist resource as much as

I would like to have asterisk pass along the caller ID phone number to a database server on a my local network (the same network that the * server resides on ) so that our customer service app. can pull up customer data automatially. Asterisk passes along caller ID to the phones fine, can someone tell me how to make it pass this info to my database server? Any suggestions would be greatly

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

Hi all, I've been searching the wiki and google for a couple of days now but cannot find any reference to a timing source on OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a cvs -q up -Pd before compiling) running like a charm on OpenBSD 3.6 Now I want to setup some IAX trunks to work and 3 friends and some meetme rooms but it looks like I need a zaptel timing source. Anyone can

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Hi all, My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls over ISDN2. Now I want to implement receiving incoming faxes into my setup so I did a google and some reading on the wiki. I got the spandsp 0.0.2pre10 package compiled and installed, patched asterisk's apps makefile and compiled * again. This all worked out fine. When integrating the RxFax into my dialplan the

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

I am looking at setting up redundant servers for my client and I know the first step in creating an auto fall back would be to have the FQDN set up in such a way that it points to both of the server IP addresses, but, how can I make it so that the phones will (with minimal delay) authenticate with tha backup server in the event of a primary server failure? Thanks, Dan

Hi, Somebody already made call recognition with database access? Depending of call's number, it access a database looking for that number. Where can i find something about this? Thanks in advance Pablo Fernandes

Ok, this is my third help plea for the day, however it's something that has been bugging me for quite some time. To put it quite simply, "Stop now" doesn't. Neither does "Restart Now" Well, ok, if I start *, and then "stop now" it does. However, after a day's calling (2000+ calls from SIP->Zap pri or zap pri->sip), it doesn't. The only way

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.

hi All, Sorry to repost but................ I have a server setup that runs sip no problem. I want to try a cisco phone. how do I a) Tell if I have skinny support loaded b) Load it onto a debian system Many thanks

Has anyone tried compiling the zaptel stuff under the 2.6.11 kernel? I get all kinds of errors when doing that. On the 2.6.10 it works fine. This is using cvs. Also, any news on when a 1.2 will be released? Or even available for download? :) -- respectfully, Joseph =============== ---------------------= ********** =

Hello! I'm curently using * with two old Teles PCI card (wich, btw, were hard to install and make good use of) with ISDN4Linux. The sound quality is simply perfect. However both dialing in and out through the ISDN line, there seems to be a _little_ bit of echo that eventually gets on your nerves ! Also the echo seems to get a _little_ bigger after a minute or so into the conversation. Now,

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan -------------- next part

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here