similar to: Asterisk + Cisco Call Manager Express

I need to pick all the Asterisk and Cisco People a little. My company has a Cisco Call Manager 3.3, configured via h323 gateways. I have remote users that I want to place a SIP Server on the external WAN and be able to connect their phones to the system and be able to get calls and call people in the office going through the Cisco Call Manager and the h323 router. My only problem is that Cisco

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

I've got an Asterisk to H323 bridge working... but I'm having a few problems.. I got everything working by setting up with the Asterisk box as a gateway in CCM. I've got two issues.. 1. If I call off net.. (Asterisk -> CCM -> Cisco 5300(I think) -> PRI) the calls will proceed.. connect, and I get about 4-5 seconds of RTP and * tells me the remote end terminated my call. I

Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks & Regards, Gurdeep

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The

*This message was transferred with a trial version of CommuniGate(tm) Pro* Does anyone have experience with connect Asterish to Cisco Call Manager 3.x? What I would like to try to do is to connect my home asterish into the office Cisco call manager most likely over a VPN from home to the office Thre may be security risks, but I'll review them as I go. Ideally, by dialing an extension in

Hi all, I want to make a SIP trunk between a Cisco 2811 router and a Asterisk. Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk 2XX). Now I want to configure a trunk so that 2811 users can call * users. I've been reading a lot but I'm still confused. Hope you can help me, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

Hi, This post may be a little off topic but I've seen lots of good ideas com from the list, so here it goes: I'm in the need of replacing Cisco Call Manager 3.2 that I have working as my primary GK. I thought I could replace it with gnugk, but it lackes some of the functionality that I need: 1.) Route retrying. If a call placed on a route fails for some reason try other routes 2.)

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

Hello I have the following structure SIP H323 (chan_h323) SIP Phone ------------------------> Asterisk/H323 -------------------------------------------> CISCO Gateway (CISCO 2610/NM2V-VIC-2BRI) -------------------------> ISDN SCCP Phone --------------------> CISCO CCM V3.3 --------------------------------------> SCCP H323 I have the

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Hi to all, I'm using asterisk 1.4.9 with chan_h323. When someone in the H323-VoIP cloud dial 1234 this number is assigned to my asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk though the dialplan can delivery the call to a particular SIP phone...this is ok... I can also dial from my sip phone every phone in the H323-VoIP cloud like siemens....BUT...when I call to a

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified

Hi I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then

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, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to