similar to: Agents, Queues, and Voicemail

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,

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

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

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

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

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

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

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

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 am having a problem with createlink not wanting to be disabled in my agents.conf file. No matter what when an agent picks up the phone, it appends the filename. Is there something other than 'createlink=no' that I should be adding to my agents.conf to prevent this? Thanks, Kyle Sexton -------------- next part -------------- An HTML attachment was scrubbed... URL:

Please see the attached file and let me know if you need any more information. /Sam Sam Sexton <mailto:sam.sexton@reuters.com> Reuters Coventry Automated Dealing Technologies Phone: +44 24 7625 6562 Fax: +44 24 7655 5203 --------------------------------------------------------------- - Visit our Internet site at http://www.reuters.com Get closer to the

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

On 2/5/2015 12:56 AM, Steffen Kaiser wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Wed, 4 Feb 2015, George Sexton wrote: > >> I'm seeing two bogus messages appearing my Maildir/cur directory. >> They're dovecot.index.log and dovecot-uidlist. >> >> -rw------- 1 gsexton users 51 Feb 4 09:04 >> Maildir/cur/dovecot-uidlist:2,S

I sent three messages to the list on Tuesday, but none of them have reached the archive, and I've not seen a reply. I'll send the other two separately. The problem may be at my end, but I didn't get any error messages. /Sam Sam Sexton <mailto:sam.sexton@reuters.com> Reuters Coventry Automated Dealing Technologies Phone: +44 24 7625 6562 Fax: +44 24 7655 5203

> I'm having an issue deploying softphones into my DUNDi/regcontext > setup. My current design is that all SIP users get registered into a > sipregistration context in the sip.conf. I then have a dialplan > function that includes that and does the dial: > > include => sipregistration > exten => _XXXX,2,Answer() > exten => _XXXX,3,Wait(1) > exten =>

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,