similar to: Re: Transfer issue with a Cisco CCM/phone (Peckham, Christopher)

Hello, We have a mixed environment here consisting of a number of Avaya PBX systems, a group of Cisco Call Managers, an H.323 gateway on a Cisco router, and an Asterisk server. The PBX land is connected to the VoIP land using the Cisco router/H.323 gateway. The Asterisk system is running code from the CVS tree from around mid Oct 2005. The OH323 driver on the system is from inaccessnetworks,

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 problem which I'm facing with CCM 5.x?

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,

The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants 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!

We are planning to install a voip system based on asterisk for 2000-3000 retail 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

I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I have just about everything working except for the message waiting indicator. I have the following setup in context [ccm] in my extensions.conf file: ;MWI exten => _2807XXX,1,SetCallerID(${EXTEN:3}) exten => _2807XXX,2,Dial(SIP/28888@65.202.115.240) exten => _2807XXX,3,Answer exten => _2807XXX,4,Wait,1

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]

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

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

Okay, 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

Hi List, I have a system that connects into Asterisk 1.4.41 using CISCO CCM 7. Everything works great except when a call is transferred to the operator. The call goes to the operator via a native bridge and is completed, then a "phantom process" starts and tries to launch a new call every 15 minutes. I modified the dialplan to hangup these phantom calls, but no still

Hi I have found in http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html (Hack to make * -> (H323) -> CCM -> IOS GW work) that i need a special version of chan_h323, because of the External RTP problem. Do you know exactly which version is it? Or do i need an unofficial patch? Thanx Andr?s

Author: Darren Moffat <darrenm at opensolaris.org> Repository: /hg/zfs-crypto/gate Latest revision: 2b29daa2195d09df740d6a13d5677c09974205e5 Total changesets: 1 Log message: last update to AES_CBC_PAD buffer size support broke CCM mode (oops!) Files: update: usr/src/uts/common/crypto/io/aes.c

Author: Darren Moffat <darrenm at opensolaris.org> Repository: /hg/zfs-crypto/gate Latest revision: d22dab8b3cf8927274becf145e811389891f4fb7 Total changesets: 1 Log message: lint nits and CCM IV/nonce size fix Files: update: usr/src/common/zfs/zcrypt_common.c update: usr/src/uts/common/fs/zfs/zio_crypt.c

I am using a CallManager Server at work and all my phones register with it. I have built an Asterisk server and am connecting to the CCM with OH323. This is working great and am going to get more features working as time permits. I have calls going both ways between these servers. The question that still keeps popping up from my users is 1. How do I get the Cisco phone to light the MWI light on

I've started to look at how I might go about using the Voicemail functionality of Asterisk in our Call Manager environment we have in the office. We're currently using Unity but want to look at moving off that if at all possible. I've read a few things on the wiki and also taken a look through the mailing list to see what info is out there to do this kind of setup, but also wanted to

Just saw this on Cisco's software download site: 7941/61 IP Phone SIP phone load - for CCM v5.0 Has anyone used this with Asterisk yet? Josh

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 licenses) , if in h323 conf i try to : disallow=all ,

I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it

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