similar to: Transfer issue with a Cisco CCM/phone

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'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 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 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! 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

Greetings list, I've been playing around this afternoon with Flash Operator Panel, trying to get it to do attended transfers. I am running the latest version. Has anyone managed to get this working reliably, and if so, would you mind sharing how you did it please? Alternatively, are there any other GUIs (free or commercial) that reliably support attended transfers? I'm trying 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

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

Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of

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 I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

Hello, ? (using trixbox with Asterisk 1.6.0.26) ? I am looking for information about how Asterisk should notify the unavailable (SIP) state of a SIP device. ? I found out that the phone (SPA504G with attendant console) sends a SUBSCRIBE request with an Accept: application/dialog-info+xml. ? The situation is that the BLF leds are green even for phones that are currently not "online".

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The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes "hold", "transfer", dials the extension and announces the call. When the attendant pushes "transfer" the second time, the original call is lost. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that

Hi all, Has anyone put * in between an Avaya and Cisco system to connect two offices together? I was thinking about adding a SIP trunk on each side and getting Asterisk to pass calls between them. There is a leased line for bandwidth. Any tips/ideas on whether this is possible or dumb? Thanks.

As I understood, Asterisk has a lot of features but lacks native 3-way calling and attended transfer. It would be great to have these features available to a simple IAX phone. I wonder how this could be implemented in Asterisk without asking for a patch. It should be possible with parking, conferencing, AGI and the manager interface. The extension 77 could be used by the attendant to blindly

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?

Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for using g729 codec. The post was

Hello all, first post, need help. I'm running a complex asterisk 1.8 install with five machines. I inherited it and don't fully understand it, nor the deep mysteries of asterisk either. I would appreciate any insight you might have. I scoured the 'net and the Digium wiki and my Google-Fu has failed me. I've been asked to somehow prohibit transfers to extension 3232. It has to be

Greetings list, Have there been any further developments recently regarding presenting the original caller's caller ID to SIP devices after an attended transfer? I've googled around on the topic, but most of the threads I've found (some from this very list) are all dated back in mid-2006 and I wondered if there have been developments on the issue. To recap, the desired behaviour