similar to: replacing cisco callmanager with asterisk?

Hi..... I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted for SIP) and a SIP softphone on a W2K box.....and it all seems to work very well.....to those who wrote this software, it is really cool. Anyway, I am new to this software, and I have a lot of questions which I am hoping someone on the mailing list might be able to answer for me.....I am basically trying to

Hi Guys I have just come from a customer that is looking to install 13 Cisco CallManagers into all their branches, (i tried to convince them to go *). They are looking for a voicemail solution. Now as Kinesis and Unity are way too expensive (apparently cisco is launching a cheap voicemail system too) I was thinking of installing * as the voicemail solution. Lots of goggling i have found

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

On 06/03/2019 19:30, Pete Geenhuizen wrote: > > > On 03/06/2019 09:59 AM, Fred Smith wrote: >> >>> I tried to install this and ran into a conflict >>> >>> --> Processing Conflict: >>> nvidia-x11-drv-340xx-340.107-2.el7_6.elrepo.x86_64 conflicts ocl-icd >>> --> Finished Dependency Resolution >>> Error: nvidia-x11-drv-340xx

Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports.... Thanks, Isaac

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

On Wed, Mar 06, 2019 at 07:26:42AM -0500, Pete Geenhuizen wrote: > > On 03/02/2019 05:41 AM, Phil Perry wrote: > > > >Hi Gerry, > > > >I've updated the nvidia 340xx legacy package set for el7.6, and > >released the packages to the elrepo testing repository. > > > >Please could you test them and let me know if they work as > >expected. To

On 06/03/2019 21:53, Pete Geenhuizen wrote: > > > On 03/06/2019 04:21 PM, Phil Perry wrote: >> On 06/03/2019 19:30, Pete Geenhuizen wrote: >>> >>> That is all good and well, but my problem is that I now can't install >>> >>> vlc smplayer mplayer, or? ffmpeg-libs because they need >>> ocl-icd-2.2.12-1.el7.x86_64from epel.

Following up on a thread that I started about Agents/Queue and acknowledging calls before bridging them... Greg Boehnlein said that he was putting his efforts into ICD. I downloaded and installed ICD, and I can get simple queue and agent stuff working fine, and see that this new design is much cleaner and more powerful. That said, in the sample conf files, the "acknowledge_call"

Check to see if you can get a IOS code leverl that supports SIP on the 6500. then maybe you can use your E1 card directly. you can also get a SIP version of the code for the 7960's etc Dave >>> jwi@weball.csis.dk 6/28/2003 2:56:12 PM >>> Hi Chris I've done a lot of things with Cisco AVVID solutions in the past. > CallManager).....am I right in saying that Cisco

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: I want to receive calls from my SIP proxy and re-route them to one of the analog lines on my Cisco VG200 ia MGCP and Asterisk. Inbound SIP calls will arrive with the five digit called number preficed with an "m" by the proxy. I'd like to them match these calls against a rule like exten => _mXXXXX,1,Dial(MGCP/aaln/1@128.100.10.11) . This however results in an error.

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 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.

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

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 new to R (and rusty on SQL!) and I'm trying to extract records from a SQL server database. I have a table of patient records (LoadPUS) which have three code columns which i want to evaluate against a list of particular codes (CVD_ICD$ table). Given the size of the patient table I want to restrict the data I pull into R to the data I only want to analyse so I am using SQL to do this.

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).

Im trying to get my asterisk box to register to a sip provider without much success. here is my console output in asterisk Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration for 'voicemail.nexband.com@metaswitch.nexband.com' timed out, trying again -- Got SIP response 403 "From: URI not recognized" back from 208.149.73.5 Urgent handler in my

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: