similar to: Asterisk and Call Manager

Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make

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

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

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

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?

Hi, Apologies if this has been asked before, but I don't seem to be able to find any info on it anywhere. Sometimes when placing a call on hold, the caller hears digitized/ robotic music on hold that gradually improves over the course of about 20 - 30 seconds until it sounds pretty normal. The first time the call is placed on hold the music sounds normal. If that same call is

Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

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

Folks! I dont know whether anyone has done this exercise before of putting together a Wish-List of things that you want to do, if you have all the gadgets you need and have a client base that needs Asterisk's Features and more. Here are some of the scenarios I am playing out that I will do, once I have enough time. Can anyone add to this list of Scenarios with or without using other gadgets,

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

More top posting goodness... Please post your updated dialplan. After making the change, did you reload/restart Asterisk so the changes would take effect? --Tim ----- "Mohit Saxena" <MohitS at starcomms.com> wrote: > Still no luck.... > > Br, > Mohit > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com >

Yes, regarding the unsigned to signed conversion you are right, it is implementation defined. I just had an issue a couple of years ago with a compiler which incorrectly treated unsigned overflow as undefined rather than implementation defined? Regarding the 64 bit profiling: I looked at the disassembly (gcc ?c ?S ?O2 ../opus/silk/sum_sqr_shift.c ?I../opus/include ?I../opus/celt) of the 64 bit

Hello! However I'm a newbie in Asterisk/VOIP/CM I would like to make sure that this system design can work: Cisco 2811 Voice Gateway - sip trunk1 - asterisk on linux box - s?p trunk2 - Cisco Call Manager 6.0 There is also a Siemens Hicom old pbx connected with QSIG to the Voice Gateway. I would like to record all calls with mixmon going through Asterisk. Is it possible? Also if there is

I'm wondering about the difference between Cisco Call Manager and SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to speak through a NAT to an asterisk box, without having to do a bunch of port forwarding on the NAT device. Without the nat, everything works fine. If the phone is behind a cisco pix that is doing the natting, it works fine (fixup protocol). If the

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-----

Hi all, We want to put an Asterisk Voicemail Server behind a Cisco Call Manager. The idea is to have Cisco Phones (SCCP) registred to the CCM and the voicemail in the Asterisk Box. The trunk inter PBX is in SIP. My question is: Is it possible to activate MWI LED from the Asterisk to the Cisco Phones registred to the CCM when they receive a new voicemail ? Thanks in advance Fred

*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