similar to: 7960 (8.2) - Call Center - REBOOT

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

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 =>

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've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you

Has anyone managed to get their hands on a 6.0 image for their 7960's yet? Or is it still in beta? Rumor (official rumor, from Cisco) is that it supports paging and intercom. I'm anxious to start working with those features, if they've been implemented sanely. What would be just as nice would be NOTIFY messages for pushing XML URL's to the phones, but sadly that feature

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

So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I have the following gripes, which I've sent to a very clueful Cisco person already. Mind you, I love the Cisco 79xx series phones, and currently they are what I recommend to anyone who wants a 'real' IP phone. I just cringe - Speed dials. It's nice to now have speed dials in the line appearances that

I have a couple of questions I'm hoping folks can help me out with. When I search the mailing list, I see folks doing what I'm interested in so here's hoping ! - How are people making out with interfacing to the 7960? I'm considering buying a number of these as they look quite feature rich. But, are they easy to interface to? - Will I be able to interface with softkeys on

I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the

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

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

I've been scratching my head for a while and I expect it is my mediocre knowledge of Asterisk which is holding me back. If anyone can assist me with some pointers I'd be grateful. Basically, I've hooked up a Viking intercom at the front door. It hooks into an fxs as a "phone". Up till now I've just played back a "go away" message if any internal phones are

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

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 apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was

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

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using