similar to: International Calls still failing - Confused!

I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for

Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring

hi. could somebody explain how exactly the following parameters in zapata.conf work: pridialplan prilocaldialplan internationalprefix nationalprefix localprefix privateprefix unknownprefix the wiki & comments doesn't quite explain them. and phone companies are absolutely no help. i've setup systems in the US & China with trial & error until it works. now i'm setting up a

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

hi folks. does any one have experience setting up E1 PRI in Shanghai, China? it works fine when we use SIP phone to dial out, however when using forward function on the same phone, it seems like it's dialing out but there's actually no respond from the phone company (China Telecom) and eventually the dial command will timed out. here's our PRI portion of zapata.conf:

And I'll resend this one too. Silly scalix. --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] zapata.conf: recent changes? Looks like you've stopped compiling libpri. All those options that are being ignored, are being ignored because they're

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the

Hi, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a "0" as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success:

I'm curious as to what versions of everything you are using. Reason being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing it to SIP/5000-00000000". It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to SIP/801-0000000c" [1-1 being the span and channel

Hi. I have a problem with two asterisk servers with version 1.2.5. In one server there is a Digium TE411P in the second the Digium TE100P. We use E1 and EuroISDN. '/etc/zaptel.conf': ----- begin ----- span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ----- end ----- /etc/asterisk/zapata.conf': ----- begin ----- [channels] pridialplan = unknown prilocaldialplan = international

Hello, all. Could you tell me how to set the type of number in the outgoing SETUP message sent over PRI trunk? I need to have: Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ] But always have: Called Number (len=18) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan

Hello, I'm having problems making international calls from our asterisk using an ISDN30 in the netherlands. Below is the zapata.conf that works for all national calls, but international calls all fail a RC=41 ; zapata.conf [trunkgroups] [channels] language=nl signalling=pri_cpe switchtype=euroisdn callerid=asreceived ;pridialplan=unknown ;prilocaldialplan=unknown immediate=no

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hi everione,<br> <br> I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the timer fail with a telco

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

back to another ? Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply