similar to: WMI from Asterisk to Cisco Call Manager

Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide

Hello, I'm trying to connect Cisco Call Manager 3.3 with Asterisk using H.323 Gateway. When I place call from a SIP phone registered at Asterisk to SCCP phone at CCM I can hear the voice in both directions. But when I call from SCCP phone at CCM to SIP phone at Asterisk the voice goes from CCM to Asterisk only. All devices have real IP-addresses - no NAT is used. Asterisk console does not

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes

Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each,

It can be done. 1. Setup a new Vm profile on CCM with a mask of XXXX 2. Setup a CTI route point: a. Set the directory number to a pattern. I use *27XX but any pattern that you can send from * is good, ie. 88XXX b. Set the VM profile to the newly created profile c. Set the line to forward all calls to VM 3. Change the dialplan in * to append the extension called to the

Hello I have the following structure SIP H323 (chan_h323) SIP Phone ------------------------> Asterisk/H323 -------------------------------------------> CISCO Gateway (CISCO 2610/NM2V-VIC-2BRI) -------------------------> ISDN SCCP Phone --------------------> CISCO CCM V3.3 --------------------------------------> SCCP H323 I have the

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's some odd way of testing if the > CallManager ("CCM") really works... >

I've got an Asterisk to H323 bridge working... but I'm having a few problems.. I got everything working by setting up with the Asterisk box as a gateway in CCM. I've got two issues.. 1. If I call off net.. (Asterisk -> CCM -> Cisco 5300(I think) -> PRI) the calls will proceed.. connect, and I get about 4-5 seconds of RTP and * tells me the remote end terminated my call. I

I've seen this question floating around, but yet to see any answers. PLEASE let me know if anyone has figured this out. I've got a SIP trunk between Cisco Call Manager 4.x (10.200.204.10) and Asterisk 1.4 (10.200.204.40). I'm trying to send calls from CM to Asterisk. It appears Asterisk is sending info back that CM doesn't like. I keep getting a SIP/2.0 400 Bad Request -

> Hrm, I made a new extension, made sure that I could dial it from a ccm phone -> > asterisk. then I set that as an alternate MWI extension ( it always put X in as an > additional entry ). the AGI I have behind that extension never runs ( in fact, > unity never calls asterisk ). > any other suggestions? You mentioned that a CCM phone can dial the Asterisk extension. Do

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

Here's the problem, I upgraded all of my 7960 phones to SIP. Now my boss wants to carry his phone with him between offices. The other office has CCM which is set up for Skinny. Now I have to put SCCP back on a 7960 phone and it won't take it. Does anyone have an example of config files for sccp. Are they the same as config files for SIP? I've never had to go back to Skinny once I

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 --

Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured

Hi all, I have a Cisco 7905g with the default cm firmware. It is connected to * using chan_sccp.so. Normal operations with this phone work perfect. Now I'm implementing a click to call app in php. This php app connects to the manager interface and sends the following down the line: $this->_write($fp, "Action: Originate"); $this->_write($fp, "Channel:

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Hi All, have some decent success on the 7920 "activation" in Asterisk. Latest status: chan_skinny does NOT work with 7920 chan_sccp does WORK with 7920 (!!) however: to remove coredumping the chan_sccp just comment out the MWI (messagewaitingindicator), then it compiles fine. Then change sccp_helper.c: return "P0060302" instead of the old value. and voila: Phone is

> > When dialling a number, my 7960 always says "External call". Is there > any chance to tell it the difference or make it display the dialled > number (especially nice when using speeddials)? > CallInfoMessage issue, it should be fixed by Jan in the CVS, I will check it out > Is there any way with chan_sccp to have asterisk taking influence on the >

Like some other people on here, I am trying to integrate Asterisk for VM with CCM version 3.x. I've got gnugk and Asterisk running, I've got CCM registering with the GK, I've got the voicemail pilot and profiles setup. A call comes into a CCM phone, it rings, rolls to the correct VM on ASterisk and asterisk emails the voicemail and I can check the voicemail, but I cannot get MWI