similar to: Re: Re: Re: Confused of London - How to associate zapchannels to extensions

----- Original Message ----- > Ooooh, I've got it working a treat, thanks to some great help from people on > this list. I *am* currently writing up a how to, and will post it as soon as > I'm done. > > Basically, take the line from the meridian, and chop the rj45 off. (!) put > another RJ45 on, but with the wires crossed (IOW a cross over cable). Plug > the meridian

I was playing around with the Flash Operator Panel, and came smack into a brick wall. We have a * box linked to a legacy Meridian System using a EuroIDSN link (TE405p) with 10 channels enabled. I also have several SIP extensions. What I wanted to do was to have a button for each of our (say) 32 users, 5 of which are on SIP. That leaves the other 27 on Zap. A potential of 27 users on 10

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? I built another route 01144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

BR C.

Trying to use IAX2 extension as call center agent but getting this on the CLI May 24 20:34:20 WARNING[1209214400]: channel.c:1783 ast_request: No channel type registered for 'IAX2[2001@2001]' using AddQueueMember as the login mechanism and that seems to work but * will not deliver to the IAX2 extension. any ideas? Jason Kawakami -------------- next part -------------- An HTML

Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

I am new to Asterisk. I am trying to ascertain the hardware setup (and associated cost) I would need. The documentation in the wiki (and elsewhere) is extensive but I am somewhat lost in product model numbers. Hence I need an initial recommandation to work on. 15 incoming lines, 25 employees). Initial scenario is to use * as a plain old PBX. I need voicemail, ability to transfer calls, ... I

I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple

is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I

Hello All, Just looking some comments from gurus about this proprietary systems and phones: Inter-Tel Eclipse2 Model name: IP PhonePlus I did not find anything useful or reasonable about their products on their website or even in Internet.... except sales. -- Thanks and regards, Vasyl Rublyov