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> Message: 1 > Date: Fri, 21 Jan 2005 17:38:27 -0600 > From: "Henry Devito" <hdevito@qwest.net> > Subject: [Asterisk-Users] SPA-2000 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn> > Content-Type: text/plain; format=flowed; chars...

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

I have searched both the wiki and googled looking for a solution to a square key configuration. I need to have C.O. lines to appear on the buttons to facilitate a small office. All of the users can see each other and calls are put on hold and picked up by the other users instead of transferred. Has anyone done this? Can it be accomplished and how is it accomplished? Thanks in advance.

...ap-time, and such? Thanks very much. This looks like an exciting project. I'm looking forward to playing with it! -- Michel R Vaillancourt Avaya Phone, UNIX Systems & Cansec Security Admin WebHelp/ QDevCo 514-295-1677 ------------------------------ From: "Henry Devito" <hdevito@qwest.net> I believe he is referring to astwind it runs on colinux that is installed on windows. This does no support any hardware yet. So without lines it is useless to me other than for demo purposes. > http://www.digium.com/index.php?menu=astwind -nathan ___________________________...

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue working ok, I have a question though. I am using Cisco 7960's with SIP for the a...

Which sound file is the one you hear when you call voicemail and it says Comedian Mail? I can't find it in the sounds directory -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050329/a007d73a/attachment.htm

Hi, Has anyone had any luck using modems on ata's other than with Cisco ATA-188's? I really don't have the money pay for the 188's as this is for my personal use. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041003/5ffde3f4/attachment.htm

Hi, I am on my next venture now, Need to set up 3 queues. I would like these setup using the agentcallbacklogin. Does anyone have an example of this? I have looked through the wiki , but all that did was confuse me. One of the problems I'm having is how do I configure my extensions.conf to dial the agentcallbacklogin -------------- next part -------------- An HTML attachment was

Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension?

Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Lots of interest here for conferencing. I've probably convinced more people to start using asterisk@home for this feature than anything else. Can I input some suggestions; Need to change the rinky dink call icons (let me know if you need some better samples) Need to change the ability for a muted caller to flag they have a question and a desire to talk (I do a lot of mute all except 1 or 2

Hi all, Happy Wednesday! Got a 7940G off eBay to tinker with and, per the wiki, contacted Cisco about a service contract. Being just little ol' me, they won't deal directly and blew me off (I hate dealing with Cisco, Nortel, etc as a small-business person...) Again per the wiki, I hit CDW.com and found CON-SNT-CP7960 for $9.14. Looking for support for the 7940 (not 7960), I just

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Hi, Has anyone had any luck setting up Cisco's soft IP phone on Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040925/7b888a60/attachment.htm

Is there any way to program a DSS or BLF key for another extension on a Cisco IP phone? I am using 7960 w /sip and a couple 7910's with skinny. I would like to see if another user is on their phone without the web interface. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have 1 phone it is a 7910 that when I try to make calls it seems to be trying to use g.723. The error I get is cannot find a path from GSM to G723. Am I missing something? Is there a way to set the codec in the phone or in the skinny.conf? I'm stumped -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can't seam to make a call transfer. I tried dialing pound and transferring a call to extension 222. This just results in a disconnect. The exact steps a took were Dial pound, dial 222 and hangup. This is the result from the verbose output. Recieved Stimulus: Transfer(0) Collected digit: [2] Collected digit: [2] Collected digit: [2] skinny_hangup(Skinny/8077@duba-2) on 8077@duba Am I

Does anyone have instructions on updating the image on a Cisco 7960? I have tried the Cisco documentation. I've upgraded them before using call manager with out a problem, but I just can't seem to get them upgraded using a windows tftp server. -------------- next part -------------- An HTML attachment was scrubbed... URL: