similar to: Why is agents.conf not utilized? (aka: can't find good info on agents and queues for AMP)

Hello all. Anybody around that is utilizing the PauseQueueMember and UnpauseQueueMember applications? Or even the AddQueueMember and RemoveQueueMember applications? I'm trying to set these applications up to function in relation to the agent number, rather than the extension the agent is at. I'm not having much luck. Anybody have any pointers or suggestions on how to get these

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

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

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

I have a call center where we're looking at converting it from a traditional PBX w/ digital phone "agent" sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad,

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

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

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Hi all, I have configured the queue below, but when I go into the queue, asterisk does not announce hold time: [support] musiconhold=>default strategy=ringall context=check_time timeout=20 wrapuptime=1 maxlen=3 announce-frequency=5 announce-holdtime=yes joinempty=no leavewhenempty=yes reportholdtime=yes I've tried changing timeout, announce-frequency, but still the same; queue works

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

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

ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?

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

I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Hi! I want to detect brute-force password hacking attacks - thus if there are too many failed login attempts for a SIP account I want to "lock" this account. Does somebody have any ideas how this could be implemented? thanks klaus

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:

Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a