similar to: OT: CallManager ANI restamp.

We've been using RNK Telecom for terminations for our SIP service, but their billing is questionable, they've been in breach of contract multiple times, and when I brought it to their attention, Russ Man, our 'friendly' account manager told me, and I quote "If you are having that much of a problem..please find another carrier." With customer service like that, I've

HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070603/6564c117/attachment.htm

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

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

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

Callmanager does nothing than construct and tear down calls and the actual RTP stream does not flow through the Callmanager but is direct from IP device to IP device. How does this work with Asterisk? I read something that lead me to believe that Asterisk has to process the entire call, is this the case? Blake Parker CCNA Network Engineer Alacare Home Health & Hospice, Inc. Email:

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

We have occasional problems with failed transfers. The PSTN call comes into Cisco Call Manager, then to asterisk over a SIP trunk and then to an asterisk controlled SIP phone. The SIP phone transfers back to a CallManager controlled SCCP phone which sometimes fails. Is there a wait to let CallManager handle the transfer instead of asterisk? I have a feeling asterisk is handling the traffic even

Hi, i'm searching and trying, but can't get it working. I'm trying to send calls from Cisco Callmanager to Asterisk with oh323 channel driver. Therefor the asterisk is defined as a H323 Gateway in the Cisco Callmanager. The Call comes from CCM to Asterisk and it works but i didn't get the called number. This is needed because i want to make Voicemailboxes. If i connect via

Hi, I am attempting to authenticate ssh access against users in active directory using winbind + pam . Unfortunately all they receive is "permission denied, please try again". A tail -f of /var/log/messages reveals : Apr 30 12:32:41 HOST sshd(pam_unix)[3011]: check pass; user unknown Apr 30 12:32:41 HOST sshd(pam_unix)[3011]: authentication failure; logname= uid=0 euid=0 tty=NODEVssh

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).

ive had enough of cisco unity and microsoft exchange and im looking for alternatives to our voip system. right now, we have 3 cisco callmanagers, 1 cisco ip icd system, and 1 cisco unity voicemail system. all phones are cisco 7940/7960's and some ata186/188's. voice gateways are cisco vg200's with pri cards (5 total). im running h323 on the gateways and phones are of course

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake

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 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?

there's this one way audio problem using h323 (CVS) with cisco callmanager? has anybody encountered this problem? oh323 works ok though... or is there any workaround for this? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030731/ea855e43/attachment.htm

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