similar to: cisco 7960 intercom

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 =>

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

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

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike

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, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

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

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.

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

On a Cisco 7960 Auto Answer is only configurable using the phone (not via TFTP), does anybody know if it is possible using sip notify or any other way but walking over to the phone?

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

Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. ------------------------------------------------------------------------ | nate carlson | natecars@natecarlson.com | http://www.natecarlson.com | | depriving some poor

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see