similar to: CallManager sip trunk - callerid name?

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

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

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

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 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

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

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

Hi guys. We have currently Asterisk CVS-v1-0-08/15/05-15:53:48 connected in SIP with a Cisco AS5300 (IOS 12.3). One PRI is connected to the Cisco gateway. The problem we have is that on incoming PSTN calls to the AS5300, relayed in SIP to Asterisk, the callerID name is not being transmitted. We received the callerID number but no name. I know we are receiving the name from the PRI in

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

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 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.

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).

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

Successfully hooked up a Cisco CallManager to a local * installation via H323. Things are working fine when dialling from CCM to *, but I'm seeing a callerid of 'root' coming up on the CCM Phone when dialling into *. Is there any way of setting this to be something else? I'm guessing the name is coming from the ID that * is running as. I'm using the chan_h323 driver and

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

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you

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

Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway via SIP to *? I've seen a few people ask and a few people that say it should work, but I've never seen an actual working config. I do a debug on our Cisco gateway and I can see the callerid name, however none of the features that should send it via SIP seem to work. Cisco docs say to use the following: