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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, For a prospective customer, I need to evaluate the cost of an Alcatel OXE IP trunking licence price. The setup would be: PSTN -----<E1>---- Alcatel ----<SIP>----Asterisk ---------- SIP Phones At the moment, only a couple of simultaneous calls are needed between Alcatel PBX and Asterisk but later, this number will grow.

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

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 all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel

Hi, I have a PRI euroisdn link between an Alcatel PBX and Asterisk. I'm having some trouble with overlap dialing. Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an Alcatel prefix of type "ARS Prof.Trg Grp Seiz.with overlap". I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which always shows

Hi there, i have a Problem with dialtone and a TE401P Card. I swear I surfed the wiki, the mailing list and google for 4 hours and did not find the solution, can you help me ? In Germany I have an E1-Line and an Alcatel 4200 PRO PBX. Without using asterisk I dial the "0" on an Alcatel Phone and have the local exchange dialtone, then I can dial. Most users do not dial en block, they

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

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

Hello, I have 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my <mailto:*@home> *@home 2.8 running on top of CentOS. Both FXO Ports are on ONE Digium card. What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my *

Hi There, I have an asterisk an a Digium 4 Port E1 Card On E1 Port No. 1 I have the Telekom PRI On E1 Port No. 2 I have an Alcatel PBX that cannot be changed So I have setup my asterisk between Alcatel and Telekom In extension.conf i configured: [telekom] exten => _9149.,1,Dial,ZAP/g2/${EXTEN}; exten => _9149.,2,Hangup This works great, all incoming calls are directly routed to alcatel

Hi guys, I need to interconnect an asterisk server (1.4.17) with an alcatel pbx, because i want that an asterisk extension can call an alcatel extension and viceversa. I need to do this throug a NT port (of the digium card) for technical reasons (linked with alcatel hardware). Card installation was fine, but i'm not sure of the configuration i choose. In /etc/misdn-init.conf i put my

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 a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.

OK! Thank you for your answer. Merci de ta r?ponse... RB g.patel@wanadoo.fr.invalid (gerard patel) on 17/10/2001 16:18:46 Veuillez r?pondre ? wine-users@winehq.com Pour : wine-users@winehq.com cc : (ccc : Roland Baudin/ALCATEL-SPACE) Objet : Re: R?f. : Re: Dead keys on Wine On Wed, 17 Oct 2001 12:36:46 +0200, Roland.Baudin@space.alcatel.fr wrote: >Ok, but isn't there any

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