similar to: Call Manager or Asterisk

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

Hello List. I am setting up asterisk as a central dialplan, voicemail and conference solution, connected to 12 Cisco 1760 Routers running Call Manager Express IOS distributed around the world. This is all done over VPN. These routers all have PSTN access in their respective country. So far all is good, and Asterisks interopability with the Cisco CME using SIP is very good, although

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

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

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Hello, , I have some of asterisk to CCM4 SIP trunk oneway audio problems. I have setup a asterisk server. It's work great and have no any problems connect to local ITSP (using SIP protocol) . But we need to build a sip trunk to another CCM4 server. The network typology like this. SIP Phone (192.168.1.100) <-----------> 192.168.1.254 (Asterisk Server) 59.124. xx.xx

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

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

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]

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

Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk. Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM. Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one

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

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,

> > 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

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake