similar to: Snom Intercom

Google search freedos rom found two source-available implementations: https://web.archive.org/web/20050104085311/http://rayer.wz.cz/romos/romose.htm https://web.archive.org/web/20040806022202/http://linux.tu-varna.acad.bg/~lig/romdsk/ On 15 January 2016 at 13:43, Sergii Kolisnyk <kolkmail at gmail.com> wrote: > As for BIOS, there are single-sector base memory (real mode) memdisk >

Hello list, this works : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT}) The phone auto-answers the call... this does not work : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Page(SIP/${SIPACCOUNT}) The phone rings and does not auto-answer the call... Can you tell me

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

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

How would I go about setting this up. I have a few 7960's with an extension set to autoanswer. How do I let all extensions answer and be active? Thanks, Will -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031110/f17a2971/attachment.htm

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

Hi there, Just wondering if there is a way to speak to the person you are transferring a call to before actually connecting them to the incoming call. E.g. "Hi, Colleague, I've got Bill from Microsoft on the line here... putting you through now" Then actually transfer the call. Does that make sense!? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077

Hey guys, I have run into one last issue before I do my full * conversion this evening. I can't seem to get paging to work. I have the chan_oss module loaded as per the wiki, and I have the following in my dial plan ;here is our intercom exten => 6000,1,Dial,console/dsp when I dial it here is the output from the console -- Executing Dial("SIP/3062-4f07",

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

I am using Console/DSP for an intercom. I want to play my MP3 collection over it when no one is using it, like when they do in the supermarket. Can anyone help me with this. Any suggestions will be appreciated. -- Christopher Dobbs

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

Nils Ohlmeier wrote: > BTW the new snom model (no availability dates yet) will have more > LED's then the existing ones. What are the most important or > interesting features (existing or not implemented yet) for the > Asterisk community for this programmable keys with LED's? The only missing feature would be an integration into the Queue management,ie. my customers use

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how

Hello Everyone, I would like to create an Intercom extension that will dial a group of extensions which are connected to SIP phones. The SIP phones are setup to auto answer a particular extension assigned to one of the lines in the phone. All phones must answer and broadcast the page message at the same time. Has anyone done this? Or should I install an overhead speaker system using the

Greetings! I was wondering if it is possible (using something like a group of Sipura SPA-841 IP phones) to have * announce information about the calling party via the SPA-841's speaker to a selected set of extensions that aren't set to "Do Not Disturb"... i.e., have * say the number, or perhaps have Festival speak the name, etc. If so, any hints and tips on how you'd go

I have the auto answer working fine on my 7960 phones on line 2, but when line 2 answers, it does not make any sound until the call begins speaking. I would like to have the phone beep when it answers so the person being called knows the speaker phone is in use. This would be much like a overhead paging systems that plays a tone before an announcement is made. How can I have asterisk answer the

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

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hi, I'm trying to integrate the following into my Asterisk environment. BPT Targa Single button audio panel (http://www.bpt.co.uk/entry-control/pictures/albums/targha/pages/HSC1ST%20Targha%20Entry%20Panel%20Mounted_JPG.htm), linked to an IT200 interface unit (http://www.norbain.co.uk/support//manuals/ref:M412CA0D98C67B/) page 6 I have been told by the engineer that the Targa will