similar to: Organization wide

Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also

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"

----- 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

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 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, How can I make calls from Asterisk client to NEC NEAX 2400 traditional phone ? Is it possible to have a connection between Asterisk and NEC NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP. Please help me to find a solution ;;; Thanks & Regards Ram Kumar Customer Support Engineer Barcode Gulf LLC Dubai , UAE Mobile : + 971 50 5594178 Email : Ramkumar@barcodegulf.net

hi * users, anyone out there ever come across, worked with or heard anything about NEC's Univerge SV7000 Telephony Server? Link is at http://www.univerge.nec.com/products/list/sv7000/sv7000.html I'm just wondering whether it's as flexible, programmable and configurable as Asterisk. It also looks like they've got a whole range of IP phones

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

OK, I think I'm ready for the next step from the X100P. Splitting a T1 with a T100P and a channel bank/s ~ 50 extensions. I think the existing system uses NEC phones for ~34 extensions, I don't know the specific model yet of the phones or the channel bank. If the channel bank is not compatible with Asterisk/digium what brand should I be looking at to interface best with Asterisk and

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

I have had the opportunity to test many IP phones in the last 6 months and I thought you might enjoy a quick review of what I have found. Grandstream Budgtone 200 - Poor Quality for business use- Looks good, and the handset feels nice, buttons have a decent feel, but the disply is difficult to read when you are not directly over head of the phone, plus the sound quality of the handset, and

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, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones,

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

I just realised I hadn't updated the version of PHP Markdown and Extra on the dingus as I pretended I did in my two announcements. It is now fixed: PHP Markdown 1.0.2b7 and PHP Markdown Extra 1.1b1 can now be tested on the dingus. <http://www.michelf.com/projects/php-markdown/dingus/> Sorry for any inconvenience. Michel Fortin michel.fortin@michelf.com http://www.michelf.com/

----- Original Message ----- <snip> > That's what I was afraid of - we have more extensions than channels ... the number of extensions isn't the issue. it is the signal the meridian broadcasts when your digital telephone on your desk is on/off hook, or ringing or whatever. if you could tap into that information and monitor it with something similar to *'s manager interface

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Many git commits in the monorepo look like the following: [Tag0][Tag1] Title line Summary: Lorem ipsum dolor sit amet, consectetur adipiscing elit. Quisque mauris neque, porta nec tristique at, sagittis vel nisi. Fusce pharetra nunc et mauris consequat venenatis. Reviewers: username0, username1 Reviewed By: username0 Subscribers: username2, username3,

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

Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have