similar to: How to read ISDN messages - URGENT!!!!

Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo

Hi. I'm looking at ordering a 30-channel ISDN connection from telia (a swedish operator) and then using a Wildcard TE110P card with that and asterisk to do IVR. Can I be certain that the TE110P card will work with that ISDN connection? A 30 channel ISDN certainly sounds like an E1 connection, but I couldn't get any clear answers from the operator if it is. Has anyone used the TE110P

Hi, I'm looking forward to install an asterisk-server to perform ISDN-to-SIP- bridging. But for the times when I'm not available via SIP I also need call- forwarding-features. Afaik it is possible to directly forward a call on ISDN instead of opening a second ISDN-channel additional to the incoming-ISDN- channel and forward the call "by hand", right? What if I need features

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file: -------------------------------- span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us

Hi, I am building for a customer an * solution that will use 2 Digium cards. 1 x TE110P (T1 ISDN PRI) 1 x TDM40B (4 analog ports, 2 for faxes, 2 for extensions) The system will be connected to the PSTN through the T1 ISDN PRI interface. All customer extensions will be SIP phones except 2 fax machines that will use the analog FXS ports on the TDM40B card. I have been investigating on this

OK I have been battling with echo problems with asterisk on ISDN for a few weeks now, and still can't solve it (although I think I have tried everything I can find.) I will try a post everything I think is possibly relevant that I can remember with the hope someone can point me in the right direction. current spec I am using (Trixbox 1.0 was still in beta at the time an didn't detect

what happens when you drop your gains? use /etc/asterisk/zaptel.conf and fiddle with tx and rx values. Works, most of the time. -----Original Message----- From: John McEntee [mailto:john@mcentee87.freeserve.co.uk] Sent: Wednesday, October 11, 2006 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Echo problems on ISDN. (mainly incoming calls) OK I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all I have already a fully operational Asterisk PBX connected only to the IP world. As a necessary step before smashing our current PBX, I need to install and configure an ISDN card able to take an incoming E1 line provided by Eircom. That's the one and only ISDN line we will ever have, so I am planning to get a single port card similar to

Hi everybody, I have a Asterisk installation working with a Digi card (HFC-S PCI A ISDN card) and another Asterisk installation with a Digium (Digium Wildcard TE110P T1/E1 Card). For Digium I used zaptel drivers but Digi works with bristuff drivers. Both installations work fine and are connected to two different phone lines. What I want to do is let calls incoming on the Digi installation be

Phil, It sounds like your carrier is just using a channel bank to split off six of your E1 channels into analog FXO ports. You will want this for fax lines, security systems, and dialup connections since doing this over VoIP/Asterisk can be problematic. It is best to keep these lines out of the PBX. I have done several implementations on IBM x305 and x306 servers and they work great. I

Hi everyone I have a Digium card TE110P and after plug it, turn on the computer and configure, the LEDs don't light up in spite of what Digium FAQ says about the LEDs: When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP? ... The TE110P LED's will light up RED when the span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is

Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_saycountpl.so] => (Say polish counting words) == Registered application 'SayCountPL' [func_cut.so] => (Cut out information from a string) == Registered custom function CUT == Registered custom

hi! I have a question of ISDN PRI "calling number" setting for outgoing call. I need the receiver to see the CLI that I have set. Eventough I've set the CLI as below, the receiver keep on getting a fixed number( used with the PRI to receive incoming calls). My ISDN PRI E1 provider says he has not done any restricions on the custom CLI side.

hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee

I am hoping someone could shed some light and point me in the right direction? I'm attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side I've searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

Hi, I've just been made aware of Asterisk through the article on Slashdot. We're a small UK company with a ISDN-30 link to the office and a Meridian (Nortel) PBX. We're expanding and moving offices soon and I'm looking seriously at VoIP technologies. So starter for ten, can Asterisk interface in with ISDN-30 technology? I'm assuming the set-up is a server sat running

Hi, I plan to install asterisk and connect it to telco through ISDN in China. I'd love to know if the ISDN standard in China has any difference than in America before I buy the digium card. anybody has experience in it? or anybody who installed asterisk with ISDN in asia can share their expierience? Or, can anybody give me some links to educate me ISDN knowledge about the difference in

Hi All, First of all, please no flames about the footer of this message. It's being inserted into all outgoing email by the server here without asking me personally first once I send out email. Since I've been very interested in setting up an Asterisk PBX + IVR + Voicemail system to support a small business and a foundation for quite some time now, in order to replace their current ISDN