similar to: Hack to make * -> (H323) -> CCM -> IOS GW work

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

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

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

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

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

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

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,

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

Hi, I know this question doesn't exactly relate to the core of this list but I thought it does relate to its "hacker spirit". Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance ? A friend of mine working for a Cisco VAR told me his colleagues couldn't make it, even for testing purpose. Do you agree ? Regards -------------- next part -------------- An

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,

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

Christopher- Nothing like defining a complicated environment. I do have some experience in this arena- but unfortunately, not with the OH323 driver- I generally stick to the Nufone driver, as I find it more reliable overall. YMMV. One thing that might help is if you could tell us if it ever worked, or if this is a new problem that's cropped up since a particular change. Still- there are

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]

In http://www.ssi.bg/~ja/dgd.txt I read: -- - key "gw" for ip_route_output used to select the right route for the gateway - key "lsrc" for ip_route_input used to find the best unicast route between this IP and the destination address (similar to output routing call but still makes the checks needed for input packet). -- Could someone please provide a couple

Hi all, I''m trying to put a linux GW running with this seput: Internet -> DSL Modem -> VLAN2 \ eth2.2 Linux > Lan eth2.3

Dear All I am using gtalk features with my own XMPP server "OpenFire" I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls from clients registered on my XMPP server to SIP devices by calling the xmpp accounts registered as clients on asterisk. So far so good. So if I want to call sip:1000 I call the xmpp account that is bound to that account in extensions.conf.