search for: mcgillivray

...to do the daily account administration if I could get an initial dump from NT, but would rather not have to make things so complicated. Any help at all would be...well, helpful. Please feel free to email me if I've been unclear about this. Thanks, Scott ---------------------------- Scott McGillivray Network Administrator Ann Arbor District Library mcgillivrays@aadl.org

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

...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 in debug, and just this line from the command line; May 5 14:30:18 NOTICE[5063]: chan_sip.c:9020 handle_request_register: Registration from 'Callum McGillivray<sip:2001@192.168.1.93>' failed for '192.168.1.90' Can someone tell me what I might be missing ? Or can someone give me a dump of their "sip_buddies" table so I can try and see what I might be doing wrong ? Thanks, Callum

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

...ly? If so, what fun surprises do I have in store as I set this up? I'd like to put this server into production, but need to make sure I can pull a backup before I move my users' data to it. I'd love to hear any suggestions.... Thanks a lot, Scott ------------------------------ Scott McGillivray Network Administrator Ann Arbor District Library mcgillivrays@aadl.org

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, 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 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, 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 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 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, 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

Hi all, I was wondering if someone could explain the relationship of Asterisk, SER & Jabber (XMPP) to me. I understand that there are facilities within Asterisk to use jabber to notify of incoming calls via XMPP clients, however I'm trying to work out exactly where the SER server would sit in all of this and what it's actual role would be. This might seem like a silly

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 -

Hi All, We have just configured our CVS Head realtime enviorment and we are now recieveing constant messages in the Asterisk output along the lines of ; SIP Seeding Peer from astdb: '2001' at 2001@192.168.1.90:9096 for 60. Can anyone explain this to me ? I have not been able to find much info on it anywhere. Cheers, Callum

Hi all, I have recently installed asterisk 1.4.1 and asterisk-addons (1.4). I have configured the extconfig.conf file and mapped sippeers to mysql as such; sippeers => mysql,asterisk,sip_buddies I have created the sip_buddies table and checked that we can login to the mysql db from the command prompt. I have also copied the res_mysql.conf.sample file as res_mysql.conf to the