similar to: Looking for Q.Sig success story

Does anyone have any input into integrating asterisk with a alcatel 4400 PBX. Acording to what i've found is that Alcatel uses R2 for E1 -- regards Vikram (http://www.vicramresearch.com)

Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Cheers, Igor Neves.

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy -------------- next part

Hi, I'm seeing something I'd like suggestions on: I have a queue with agents that log in using agentcallbacklogin. The extension that is logged in with is a Local channel. Now, if a call comes in to the queue and is handled by an agent (in our case using Cisco 7960 SIP phones) and transferred (attended) to another extension, the agent remains unavailable during the remains of the call.

Hi guys, Recentlly i did a asterisk gateway and use it with an alcatel pabx. All is working, i have only two problems. 1. When call incomming to asterisk, it forward to digium card to PABX Alcatel. The user that start the call can't hear the control tone of ring ring ring. Tha calls stay without sound until the called part answer the call. At this point, conversation follow normaly. 2. When

Hello, I am using a Cisco-2,811 as a gateway between the Asterisk PBX and our Nortel TX-1 university's PBX. It is working but no names are exchanged. From the debug mode I see that the Cisco sends the display name (which does not appear on the Nortel's phones) and the Nortel does not bother to send it at all. I recall that when I had a pilot with Cisco CCM two years ago we had to set

Hi all, I'm new in the forum, and although I have some experience in Asterisk, I've never work with Asterisk FXO, FXS, E1... cards. I have several costumers with ATAs working with my SER. However one of them bought an Alcatel PBX OmniPCX Enterprise and now he wants I give him a E1 interface for interconection with its new PBX. I understand I need a E1-IP gateway which could be Asterisk

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus