similar to: Terminating VOIP calls on EuroISDN PRI interface ?

Hi, I'd like to terminate IAX call on PRI interface. What conditions should be met to be able to send arbitrary caller numbers to called party, so he can call back to supplied ISDN number (that is different for every IAX user) - not through PRI interface, but plain ISDN call !! Thanks in advance, regards, Rob.

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.

Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN <---1---> TE - * - NT <--2--> Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable

> Date: Thu, 23 Jun 2005 08:50:50 +0200 > From: "Robert Rozman" <rozman@fri.uni-lj.si> > Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * > - Euroisdn Italy > > I'm pulling my hair down and getting bold :-) ..... I have Asterisk between > Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff > Asterisk).... Plenty of

Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same

Hi, I'm pulling my hair down and getting bold :-) ..... I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk).... I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for

Hi all, My problem boils down to these errors: ... Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time This is triggered by lines in extentions.conf such as: exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W) The system is CentOS v5.2 with Asterisk 1.4.23 (druid-asterisk-1.4.23.1-2), a Sangoma A104

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 a problem with setting outgoing caller id to "nothing" (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried: exten => _*70X.,1,Set(CALLERID(name)="") exten =>

Has anyone got a working zaptel.conf and zapata.conf for a Digium Wildcard TE110P T1/E1 Card. It's connected to a BT ISDN PRI (EuroISDN) with 24 channels. Inbound works fine, but outbound isn't setting CLI (it seems the line supports 6 digit CLI). Inbound CLI works fine. In the dial-plan using Set(CALLERID(num)=123456) then Dial(Zap/g1/01234567||frT) Where 123456 is in the range of BT

Hi all, finally yesterday I was able to set up my Asterisk machine and now it works greatly. Assuming I use Asterisk also to manage a BRI via a HFC based ISDN card, I can dial outgoing calls via ISDN but I still can't answer incoming calls from that BRI. I read that special extension 's' is for analog only. What do I have to put in extensions.conf to make Asterisk answer incoming

Hi, We did an interconnection with our carrier few days ago. But, I noticed that there was a signaling problem on our trnuk. In fact, Asterisk indicates that the call is answered when we received ALTERTING message from our carrier. This is PRI debug logs : -- Executing Dial("IAX2/0w955897@0w955897/5", "ZAP/g1/0130450836") in new stack -- Making new call for cr 32784

Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow] # # The timing

Dear All, I'm having a problem with my Asterisk + E100P Installation in UK (BT PRI). The system functions as expected, and my dial plan works as expected. 30 minutes (or so) after starting the asterisk service I lose the PRI line, and only get this back after a service asterisk restart or reboot. During the failure there is no alarm on zttool, ztcfg show all 31 lines and there are no

Hi, I'm a newbee to all this asterisk stuff, and after reading a fair amount of docs at the voip-info wiki, I was wondering if it will be possible to work out something as the following scheme: A B +-----+ +--------+ +----------+ +---------+ | SIP | | | Internet | PC | phone line | classic | |

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain