similar to: Transfer BACK to CallManager over SIP trunk?

Hello all We succesfully added a H323 Gateway to our CallManager 3.0 that resides in Mexico and were/are able to make calls from CallManager SCCP phones to the Asterisk Server phones in the U.S.; however, we have not been able to call from Asterisk server in U.S. to CallManager phones in Mexico Here is what we tried: 1. Adding a Gatekeeper into CallManager and then have Asterisk (and also

I have been unable to get callerid name passed from Cisco Callmanager over a SIP trunk to Asterisk. Only the number is displayed. Has anyone been successful getting callerid name?

hello all- Long time listener, first time caller. This is a great list and has given me tons of help as I've set up * for the first time. I've got an asterisk system up and running at a new company, and it does about 99% of what we need it to do. TelephonyWare has been our equipment supplier, and has been great with support, but I've got an issue that has us both stumped.

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

Hi all. I compiled the module chan_sccp, now its possible deploy trunk SCCP with Callmanager? Anyone? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120918/881c4e7b/attachment.htm>

Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone.

Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got

Hi all, I'm writing my graduation theses : analysis VO-IP protocols , and I cannot find any documents about Cisko Skinny Client Control Protocol. I have Cisco CallManager and some IP-phone and I'm sniffing traffic between that, but I don't understand, how this protocol works. Clearly i'm looking for description of SCCP commands and explanation some basic SCCP scenarios or what

I have a SIP trunk built between a Cisco CallManager version 8. I can dial the phones registered to the Asterisk PBX from a phone registered to the Call Manager. I've tried to keep the config as small as possible to help the troubleshooting process. Attached is he most recent debug. My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251 SIP.CONF [6001] type=friend

To be clear I meant using Chan)_h323 with Call Manager where CM is configured with * as a H.323 gateway, not client. CM supports H.323 to direct calls through gateways, and in fact until recently that is all they supported. They now also have MGCP, but only to their IOS platforms, and SIP is coming soon. There are NO sccp-based gateways, from Cisco anyways. Dan -----Original Message-----

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux

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

Hi list! I'm trying to configure the Kirk IP 600 wireless (DECT) system. The wireless phones are regular DECT phones, the Kirk IP 600 is doing the voip part by registering/gatewaying the phones to a callmanager server. The phones do not work and I think the problem is that they do not register at asterisk but I'm not sure because I don't understand a thing of the sccp/skinny

Hi, I setup a SIP trunk between asterisk and Cisco CallManager according the wiki page. http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration But I'm getting a 'Malformed/Missing URL' from the CallManager. Does anyone know what went wrong here? I'm running asterisk CVS HEAD and (192.168.1.5 five) Cisco Callmanager 4.0(2a) (192.168.1.101) below is the debug

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.

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

Hi, ? I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a "sip show peers" on the Asterisk server?I see the Callmanager as Monitored and online however I can't get any calls

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

I have searched and searched for over a week on this but can't seem to find the issue. Calls from CallManager to Asterisk are being disconnected immediately. I have setup CallManager and Asterisk per Shaun Ewing's pdf http://asterisk.edropbox.net/ccmasteriskvm.pdf I have installed Asterisk 1.4.0-beta3 on Fedora Core 5. I got libpri, zaptel, and asterisk compiled and installed.

I have been trying for sometime to upgrade this phone to either sip, or the latest sccp (at this point I don't care) with no success. It has an sccp image (don't know the version off the top of my head) that is incompatible with chan_sccp. When I boot up the phone, it asks the tfpt server for the .tlv file, then the SEP<mac> file which it receives successfully. It then asks for