similar to: CallerID from cell phone not being rewritten

Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg -- Kyle Sexton

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) -- Kyle Sexton

Before I head down the path of converting voicemail to an ODBC backend, I have a couple questions that I was hoping someone would know. 1. Is the voicemail message stored in the datbase, or just it's location/filename? 2. Does MWI propagate when using an ODBC backend? 3. If it does both of those things, wouldn't it work well for a centralized voicemail system instead of a solution like

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the

Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be "stuck" as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running

I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I know that I could setup simple SIP->SIP to allow everyone to call each other, but that's not generally too fun. -- Kyle Sexton

All, Is razor only really useful for configuring mass numbers of hosts? I''ve been playing with it and one thing I''m not grasping is how to set a hostname if I''m only building one host, say wiki01.foo.com. For each server like that would I need a separate policy, or is there some other way people are using to pre-populate specific host information to be passed to

All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. The problem is that since the incoming context is the same context as the normal incoming call context,

Err, stupid me. Seems like the phone is hungup, but when I start ztmonitor, the line is picked up. I guess the line in India is pretty loud b/c the RX bounces off the charts, changed it to (-4) and all my problems are solved. Regards, Min Chang On 4/14/06, Kyle Sexton <ks@mocker.org> wrote: > > Have you tried putting a Hangup in your extensions.conf? > > > > On 4/13/06,

Hi, I am running a call center based on Asterisk and building some statistics based on the queue_log file. I have some doubts about it that maybe you could help (actually, maybe these doubts are suggestions for enhancements!): 1st Scenario - Agent receives the call, and puts it on parking for somebody else to pick it up. Parking # 7000 (for attender transfer)

The language docs have been languishing for ages, so out of disgust I consolidated the LanguageStructures and LanguageTutorial pages into a new, rewritten LanguageTutorial[1] page. Let''s consider this page the single page for all language explanation, and let''s do what we can to enhance and correct this page over time. I think the existence of two pages confused

Dear Wayne, C Sights, and list, I have started working on my rewritten version of the rsync man page. So far, I have redone the top of the man page and the list of options; I improved many of the one-line option descriptions. The yodl and man versions of my man page are available here and here: http://www.kepreon.com/~matt/private/rsync-new-manpage/rsync.yo

Created this function on one of my machines today, thought others might find it useful: freesip() { comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr -d \]) | grep ^[[:digit:]] } On RedHat/CentOS based systems you can create the following file to have the function available on login: /etc/profile.d/freesip.sh # Free SIP extensions freesip() { comm -2 <(seq $2

Hi all I have my asterisk connected to different SIP Providers. I had to idea to create something like a Helpdesk queue ringing all my friends connected to those SIP providers. Now my problem is, that I have to set the correct callerID if a call goes out to one of those sip providers. I can set the callerid when the call comes in, but then this caller id is valid for all 'lines' that

We need to make sure that all queue applications run on the correct system that the user agents that own the queue application are registered to. So when a server fails and the user agents register with their secondary server (which will always be configured to be the same server for those related agents) the queue application is running on that server and routed to correctly by it's peers.

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Hi, To try to bring our wiki 'up to date' (as in, no longer talking about CVS), I've gone through and rewritten a lot of the wiki content for our main landing pages: our front page, the software and specifications pages, the infrastructure pages, and even (to an extent) the mission statement. A lot of this is trying to describe how we've interpreted these things over the past ten

All, I am experiencing an issue where if an agent is logged into the queue, but has their client closed. It appears that when the queue calls the agent, it goes through the macro I have setup for that user and will dump them to voicemail if unavailable. This pulls the call out of the queue, which is not what I would like to happen. I am wondering if this is the expected behavior and I should