similar to: createlink option in agents.conf can't be disabled?

Hi, Asterisk version : 1.2.7.1 stable version We try agent.conf setting of createlink=yes We always can not see this link value to be filled in MySQL's table filed : userfield But we can see the record file has been created correctly. In debug mode, no userfiled shown in SQL command, May 17 18:10:51 DEBUG[2889] pbx.c: Function result is '"unknown" <2001>' May

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

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

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,

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

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

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

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

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

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

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

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

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,

Hello everyone, I can not get the name of the recoding file of agents calls. I set agents.conf as following: ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ; The format to be used to record the calls (wav, gsm, wav49) ; By default its "wav". ;recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By

I have two servers setup to do DUNDi lookups against each other. The scenario is that on server A, I have a wildcard match for extensions 64XX that rings to a local extension on the server. On server B I have a 6442 real extension that I would like to have ring if called. It seems that DUNDi is matching on the 64XX and not searching out to see if there is a *more* exact match than the pattern

www.iaxtel.com seems to be down, does anyone know if there is another way to register new numbers with them? -- Kyle Sexton -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070719/1d4f998f/attachment.htm

Considering getting the card pictured here, http://www.voipsupply.com/product_info.php?products_id=2419&searchid=341962, and was wondering if anyone had any experience with those in a 2950's case. From the picture it looks to be a pretty large card. I'd hate to get it and have it not fit! -- Kyle Sexton

I'm setting rsync up for the first time and would prefer to have the exclude file defined in the conf file, but the exclusions aren't honoured when I define the parameter in rsyncd.conf - although they are when I specify the file in an argument. The server is the remote system and both rsyncd.conf and the exclude file are the same on both local and remote systems. I'm attaching the

All, Starting this month will be the first official meeting of the Kansas City Asterisk User Group (http://www.kcaug.net). The location of the meeting will be at the Daily Dose Bar & Coffee House, in Overland Park. Meetings will be held on the last Sunday of every month (the 27th for this month) at 2pm. If you have ideas or topics you'd like to discuss please let me know and we can