similar to: Asterisk + NEC Electra Elite IPK Integration

I have an nec electra elite 192 with a t1 card; and am looking for suggestions as to integrating them (can't throw out the system yet!). I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart signaling, and 2 working bt102 grandstream ip phones (thanks again Matt for your "start from scratch"

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

i am just curious how many * systems are in the real world with more than one user. do you run a certain version? you dont update CVS do you? any admins running a system of over twenty? over fifty? over one-hundred? i deal with 3com and nec systems all day (i am cerified in the 3com nbx advanced network telephony, elite voice mail, CCNA, A+, nec ipk, and soon to be "asterisk school of

Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

I recently posted a message about an interesting dilemma which I could not figure out. After doing a whole lot more digging I think I have figured out part of what is going on. The parking app is implemented as an app, ie can be called from extensions.conf) but if you look really closely, it is also implemented in each channels driver (handler, whatever you want to call it) and those

I am currently attempting to get Asterisk working with our existing NEC IPK 192 system. The * box has a Wildcard X100P clone card in it and is hooked to an analog port on the NEC system. The analog port is shared between an elevator phone and a credit card machine. I have one SIP extension configured in * which the softphone on my Windows PC is registered to. With the softphone I am able to

Is there a way to ring selective line on multi-line phone. For example if I'm on the phone talking internally on line 1 and the calls comes-in the line 2 will automatically ring. The phone P104 allow extension to be assign each line. Is there a way to call certain line (example line 3) on multi-line phone instead of line 1 when the phone is not busy? For example the Sip phone P104 has

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Sorry if this is the wrong list... I need a toll-free number to be delivered to me on IAX. (This is NOT an existing number need to buy the whole service.) Anyone know of a service provider offering this? -Mark 707-735-1038

If I have different opening hours on different days, can I accomodate that in a single gotoiftime, or will I need to filter them out one by one ? For example, our hours are Mon-Fri 9:00-17:00 and Sat 09:00-13:00 can this be done something like GotoIfTime([9:00-17:00|mon-fri][9:00-13:00|sat]|*|*?open,s,1) or something like that, or do I have to do:

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999%

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris

Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Has anyone else noticed this? I use 'set verbose 25' (insane, but I want to see *everything* right now) and would like to do the same for 'set debug', but as you see, set debug has a bad impact on my CLI output. asterisk*CLI> -- Executing Answer("SIP/824-b4ff", "") in new stack -- Executing Wait("SIP/824-b4ff", "0.5") in new

Converted a strictly VOIP system in NYC to NEC IPK TDM system... will have 25 Polycom 501's for sale. Best offer, offlist only please. R