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I've got problem with connecting asterisk to CCM. Our side has Asterisk system other side CCM , ehrn i dial a number on other side channles created , connections established but nothing happend , just silence , and after some time busy tone. Sides sending ad reciving g711 codec , but we need that sides send and recive g729 (we have license...

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 the call to an extension on Asterisk. For example, extension 7443 would forward to 27443 on Asterisk which looked something like: exten => _27XXX,1,Voicemail(u${EXTEN:1}) Obviously setting this fo...

...server callout from 077777777 to 0888888888 I then originate a call via AMI to Local/077777777 at outgoing-originate with context set to outgoing-originate-rec and extension to 0888888888. The result will be something like this: -- Executing [s at macro-ccdev2-rec:1] MixMonitor("SIP/upps-ccm-tq01-0000003f", "cbrec-15605.wav,b") in new stack == Begin MixMonitor Recording SIP/upps-ccm-tq01-0000003f -- Executing [h at outgoing-originate-rec:1] AGI("SIP/upps-ccm-tq01-0000003e", "agi://l4574/ajpbxtest.agi?path=uploadrec&callid=15605") in new stac...

Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the probl...

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...

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 look at my CCM CDR's and the call had a zero length duration....

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

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] <--ski...

...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 this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at le...

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 ast...

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 dialpla...

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this subject. So please if anyone has any...

..., another Cisco related issue (sorry!). Single Asterisk box at location 1. Single Cisco box at location 2, however the Cisco is also the PBX for location 3 (same physical machine, calls routed via VoIP). Trying to have Asterisk be able to call EITHER Call Manager location. The single SIP trunk in CCM (version 6.1 mind you) only allows a single device pool to be selected. So configuring calls to one location...no problem. One location at a time that is. If I want to be able to call the 3rd location I have to select a separate device pool and restart the trunk. Then calls to the 2nd location sto...

...nts of the H.323 channel in Asterisk use non-current OpenH323 versions, it may not be of any benefit to anyone anytime soon if I went that route! I've not checked newer OpenH323 source to see if the hack below can be shifted into the application either. I've been messing with the companys CCM PBX, which has an IOS based gateway box with a PRI behind it, and talking to it from my * box with H.323. For the most part it works. It's in Lonodn UK, I'm (currently) in Boston, MA, USA. It's all VPNed. People being able to call me has made working remotely that much easier. However,...

...locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones -> SER -> * -> CCM/Voice Gateway -> PSTN I'm very new to * and voip an so I have a couple question to get this done: 1. Is asterisk powerful enough to handle such a lot of sip users? I heard about using * together with SER. SER as the SIP Proxy/Registrar and * as a g...

> > 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 fr...

Hi, I have used basic Asterisk as a PBX controlling few extensions. My question is simple, at work there is an existing Call Manager/PBX and all which manages all the extensions for SCCP VOIP phones. Can Asterisk be used to manage just 1 VOIP phone and still can make internal calls to the other extensions? Thanks, Jorge -------------- next part -------------- An

...t up some debugs on your gatekeeper, to see if the call is being signalled properly to Asterisk, and on Asterisk itself, to see what's being passed in. This is critical- I'm betting this is a simple case of dialplan mangling- but only the debug logs will tell. Secondly, I know that in the CCM trunk definitions, it's important to ensure the trunk definition follows the recommendations *exactly*. A big killer here is the 'Media Termination Point Required' box- generally, for transfers, you need one- and it needs to be functional. Your CCM administrator should be able to veri...

...like. I keep getting a SIP/2.0 400 Bad Request - 'Malformed/Missing URL' error. What is Call Manager looking for!!!??? Here is my CM log: //////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////// 10/05/2007 15:30:17.381 CCM|Incoming UDP SIP message size 494 from 10.200.204.40:[5060]: OPTIONS sip:10.200.204.10 SIP/2.0 Via: SIP/2.0/UDP 10.200.204.40:5060;branch=z9hG4bK01e1b958;rport From: "asterisk" <sip:asterisk at 10.200.204.40>;tag=as7d7e63d3 To: <sip:10.200.204.10> Contact: <sip:asteris...

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