similar to: Asterisk, SER & Jabber

Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd like to be able to query one in the dial plan and use the results to tamper with call priorities / CLID etc. If someone could point me to a howto / guide or relate their experiences with this, that would be great ! Thanking you all in advance. Callum P.S. Before anyone asks; it's a legacy database that

Hi all, Has anyone had experience with installing AMP on a soekris box running Astlinux? Is it possible ? Cheers, Callum

Hi all, Just a quick query about something that my manager has asked about. While I have talked with Asterisk people about this before, I forget the actual answer. We currently have an ISDN 30 installed on a Dell 2850, using a single port Digium ISDN card. We would like to add an analogue card, plug our fax machine in and have Asterisk simply detect the fax and pass it through to the fax

Hi all, Can someone point me to some information on the type of hardware that might / should be used for a high load on an asterisk machine ? I know that this is dependant on what services you plan to have running, and it's relevant to what you plan to do. We are likely to be running 4 E1's, Voicemail, IVR menus, Music on Hold, Pay-Over-The-Phone, lots of interdepartmental

Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because

Hi all, We have just installed a * server (CVS head) with a TE110P card and a IDSN20 line, we are using the GXP-2000 handsets running the latest firmware (.9). Some of the calls we are receiving have echo at the our end (we can hear ourselves speak). We have a traditional ISDN telephone system here as well and when I make a call from one of the handsets asterisk answers, directs the call to

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

Hey all, My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? As I said, probably blindingly obvious. but my techies have gone home for the evening and I was looking for an answer before I left. Thanks, Callum -------------- next

Hi all, I was wondering if there was a way to deal with "wrap up" time for agents slightly better than we do it at the moment. At the moment, we set a wrap up time of 20 seconds for each of our call queues. The problem with this is that sometimes it's either too long or two short. I would essentially like to have all agents put into wrap up straight after a call, and have to

Hi, I was wondering if anyone had any problems with the Dell 1800 series servers, with TDM400 cards? I saw that digium seems to recommend against a lot of the dells. Please let me know Thanks Sascha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d8219499/attachment.htm

Hi, all Where can I get "Asterisk Developer's PCI Kit" in Australia? It is a TDM400P with 1FXS and 1 FXO module. I amight need an extra FXS module as well. Thanks, Rudolf

Hi All, We have been playing around with call queues and asterisk and now have everything set up the way that we want it, bar 1 thing. When we have a scenario of an agent logged into several queues, we want to prioritize the "queue" so that calls in that queue are answered before all the rest. While we can prioritize calls within a queue, this seems to have no affect when an agent

Hi all, Having an issue in getting our first asterisk server recognising the channels on our E1. At the moment, when we start up the machine, everything goes well, but the light on our single span TE110P card flashes red (slowly). I imagine that I'm supposed to get a green light when a connection has been made. (lol - else I'm not sure what it's there for :P) Can anyone

Excuse my ignorance (I just got crap on the #centos IRC channel for this question), but is there a (easy!) way to have and IRC and/or Jabber server relay a login to a Microsoft Active Directory server for authentication? If there's a better question to ask this question, please point me in that direction, and I'll be happy to do so Thanks -------------- next part -------------- An HTML

Hi, I'm trying to get our TE110P card up and running on our * test server. Initially we were getting a flashing red light on the card, however after changing to a crossover cable, we now get a green light for a moment, followed by a solid red light. I am also getting the following message at the command prompt; PRI: !! Got S-frame while link down == Primary D-Channel on span 1

Hi, I'm trying to compile vmkit but am running into some problems. I followed the instructions at http://vmkit.llvm.org/get_started.html and checked out llvm and clang from svn, compiled them no problem and the same with classpath 0.97.2. I then checked out vmkit from svn and tried to build it. All these are located at /home/callum/vmkit/{llvm,classpath-0.97.2,vmkit}. I used this to

I am trying to setup a local Jabberd server for our members of our IT department to use. I downloaded the source for 2.0s8 from the jabbered site. When i run ./configure I receive the following error: "configure: error: Libidn >= 0.3.0 not found" I went out to dag and downloaded and installed the libidn-0.4.6-1.1.el3.dag rpm. When i ran ./configure again it failed again. After a

Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and video. So I can establish a voip + video connection *one-to-one* only....it works OK. But I'd like to implement a videoconference *one-to-many* in order to intercommunicate many clients, is it possible with Asterisk 1.4 ??? (multicast is better than brodcast in this situation of course) Thanks a lot, Alejandro

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

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