similar to: A TDM issue..

I can make them logon but how do I make it so if I dial a number it will make the other one ring, is it under the sip config? what shouuld I include so for instance if I dialed 1111 it would ring phone 1 or 2222 would ring phone 2. thanks -chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot

I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom

Is it at all possible to set a Global Variable freely whenever a context gets used without having to enter an extension priority to use SetGlobalVar? This is really limiting the dialplan for me. Heres an example of what I would like to be able to do. [globals] AREACODE= [local] exten=_NXXXXXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah) [anyoldcontext1] AREACODE=313 include=local [anyoldcontext2]

There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris.

I appologize if this was already answered somwhere on http://www.voip-info.org/wiki-Asterisk, I'm sure it probably is. And if you wish to just point me to a link that would be appreciated. I am very new to asterisk and unix all around, so these questions may sound rather ignorant. First being, how do I setup asterisk to point to another asterisk server and make all the lines which should

Hi All... I'm trying to get the intercom feature working on some snom 190 phones but having no luck... As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. I've email'd snom a few days ago but have yet to get a response. On my 190s, im running snom190-SIP 3.57v. I am pulling the config for the

Dear members of the list, a customer of mine has some questions and I would like to pose some of them further to you guys. What is the situation with Asterisk and fax over IP ? Can Asterisk receive a fax over a POTS or ISDN line ?? Do I then need a Digium TDM-card and an FXO-module or a T38-gateway ? What phones should I use to automatically configure them from a central place via HTTP or FTP ?

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has

Hello all, I noticed that Cisco claims that you can do station to station intercom with the 7940/7960 phones, and the Cisco Call Manager. Does anyone have an example SIP header that shows this in action? Or is it something else that triggers the intercom? I would like to add this in to *. Thanks, Mike

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson

I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a

I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and

I read about this intercom stuff on page 62 & 63 of the book "Developing Cisco IP Phone Services" isbn 1-58705-060-9. Primary calls take place on streaming channel 0. When streaming channel 0 is not in use, streaming channel 1 can be used for asynchronously streaming (in and out) stuff like voicemail, email, and, yep the one we want, intercom. Page 87-88 of the book talks about

Hi, I have asterisk running a single T1 card with a connection to a nortel CS 1000. All calls to extensions, local and long distance are working just fine. My issue is this: The nortel CS 1000 supports connections to and intercom system that is just line level audio to speakers. When my PRI tries to call this "extension" it is never answered so my AGI never runs to play a message