similar to: Alcatel OmniPCX Enterprise + Asterisk with E1

Greetings list, I've been asked by someone to help them set up a SIP link between an asterisk system and an Alcatel OmniPCX (v6 software). The asterisk bit's fine, but I know nothing about the Alcatel except that it does apparently allow the setup of SIP trunks. Does anyone have experience with the Alcatel unit? Any obvious pitfalls to watch out for? Any suggestions gratefully

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

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

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

Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) --

Morning Asterikians, I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the

Hi everyone I have successfully compiled and installed OH323 support (finally) into my Asterisk. I want to connect the Asterisk server to our Alcatel OmniPCX Office (OXO) PABX, which has an internal H.323 gateway. I have created the correct dialplans in Asterisk and same in OXO. The OXO only supports G711a G711u G729 and G723.1 codecs. When I call from a SIP phone to OXO using my

Hello, I have the following setup: (*)<--->IP<--->Micronet 5012 H.323 box <---> POTS <---> PBX (Alcatel OmniPCX) Grand idea is to use the micronet's POTS interfaces to connect SIP phones to the PBX and to the PSTN. I think i even managed my way in the arcane and cryptic management interface of that appliance, but I am stuck against theese messages: -- Executing

Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm trying to connect my Asterisk@Home to iptel.org, but the only I get is Allison telling me "circuit busy now, please call again later" or some thing similar. I'm trying make it by AMP and editing sip.conf and extension.conf, and I read all about it in voip-info.org. I will appreciate your help, Thanks in advance, Sebastian e-mail:smilioto@GMAIL.com IM:

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Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

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 list, any one can let me his config files for interconecting a Meridian Op 11 and Asterisk via a E1 PRI CARD. Actually i need the nortell config part, becouse my client nortell provider doesn't know how to config the PRI card at his part. Thanks all. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with

Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it. The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware