similar to: PBX --- > Asterisk Connection

i have a traditional pbx attached to one line of NT1, and asterisk with hfc-s to the other one; when a call comes in, it is like asterisk captures it, passing it to the channel configured on dialplan; in the facts, the call is not answered, but NT1 doesn't ring pbx, as it would do if call had been answered. What is happening? Can i avoid this and have pbx and asterisk ringing

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Hello, I have the following problem with HFC card and Alcatel PBX 4400: - I want to connect Alcatel PBX and Asterisk with HFC card (zaphfc driver 1port S0/T0 interface card) - BRI line is configured at PBX to T0 trunk - In asterisk I have loaded zahfc driver in NT mode and also set in zapata.conf NT mode, point-to-multipoint. - I am using cross "isdn" cable to PBX -

Hi, I've solved the flash transfer problem changing the flash time in the zapata.conf file, I've set: flash = 200 (the defualt was 750 ms) in the extensions.conf the code is for example: exten => 42,1,Flash() exten => 42,2,SendDTMF(42,250) exten => 42,3,Hangup() now the transfer with flash works correctly. About the question whether my PBX expects a hook flash for

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

Hi list, we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra). People from telco told that they can't connect two PBX's using E1/T1 or only with QSig signaling. I wanted to use EuroISDN. In this case, it was me told that VN6-VN7 would be used. The PBX has a spare ADQ card installed on which we would connect. Has someone a such working setup? Is it working well? More

Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye

Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate("unknown", "unknown") does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa'

Klaus-Peter (while hardly able to see through my watery eyes), You've earned yourself a (potentially rackmountable) 19" Dutch cheese as soon as the zaptel driver is finished and tested. In fact, I don't even have to mail it, I can pratically bring it over ! Although I'll be monitoring your site on a 24hrs/day basis from now on, please keep me posted on both the driver and the 4

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Hi, I have received some replies for my previous mail (* configuration), asking for my goals in configuring Asterisk. So here they are: We are planning to host an Inter continental virtual PBX service that will enable our users to register for an account which give them a toll-free # or a DID. Once registered using a web based interface, that user can add as manay extensions he/she wants

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of

> -----Original Message----- > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steven > Sent: Thursday, January 12, 2006 4:37 AM > > This is supposedly fixed in 1.2.1. > The issue is that some companies using PRI are starting their > IVR as the ringback tone. > The problem is that asterisk considers the call not answered > yet, so will not send any DTMF

Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards _________________________________________________________________ Biography of Shah Rukh. His profile, awards, films.

Hi, I successfully installed 2 avm card in my asterisk box but I'm unable to make call. My capi.conf is: msn=0721111,07211115 incomingmsn=* controller=1,2 softdtmf=1 context=default echocancel=yes callgroup=1 devices=2,2 my capi info : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. my extensions.conf : exten =>