similar to: re: Restoring Cisco 7960 to defaults

Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it.

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's some odd way of testing if the > CallManager ("CCM") really works... >

On Wed, 23 Jul 2003, Yifang Dai wrote: > I wish! My company just spend a lot $$ on the shinny CCM phone system, so I > don't think I can change that easily... But if I can get asterisk to > talk to CCM via h323, and prove it's usefulness, I might have a chance > to use * in the branches... Well, good luck, then! > By the way, do you know if we can get *'s VM to

I'm trying to set-up Asterisk server and I would like to buy 2 SIP phones. Has anybody tried Cisco 7960G? Or 7940? What audio compressions can I use with this phone and Asterisk? Reason why I'm asking is because Cisco supports G.711 and G.729a audio compression (probobaly some tohers but they are not listed on data sheet) and on Asterisk features i found that it supports G.729 but need

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Hi ! We have Cisco 7912 phones, and the doc says that I can "create up to four speed dial buttons on my phone using the Cisco CallManager". Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D

Hi All, have some decent success on the 7920 "activation" in Asterisk. Latest status: chan_skinny does NOT work with 7920 chan_sccp does WORK with 7920 (!!) however: to remove coredumping the chan_sccp just comment out the MWI (messagewaitingindicator), then it compiles fine. Then change sccp_helper.c: return "P0060302" instead of the old value. and voila: Phone is

Hi All! First, let me introduce myself, as this is my first post to the list (I've been lurking for quite some time now). My name is Matt Hardeman, and I work for a software development firm in Birmingham, AL. We are interested in the Asterisk PBX and it's various configurations first as an internal solution for our occasionally bizarre telephony needs, and eventually are interested

I've got a question regarding Cisco IP Phones and licencing. When using a third party PBX like asterisk is a licence required for the Cisco phones ? Has anyone got anything in writing from Cisco to clarify this ? Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm not using Cisco Callmanager ? HYPERLINK

Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put "P0S3-04-4-00" in the OS79XX.TXT file, the phone downloads this

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

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

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Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike

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