similar to: zaphfc errors

Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error

Hi Is anyone having this weird trouble with bristuff....calls don't get answered, but on the debug it looks as if the call is answered. The call will just hear ringing. I am using -BRI-stuffed-0.1.0-RC4a Occassionally the system starts working on its own, other times I have to stop and rmmod everything and start them all up again. These error messages appear on the CLI: == Primary

Hi to everyone! I am new to this list and i want to greet you all! I am trying to install Asterisk on a Fedora Core 3 Distribution with a Digi Datafire Micro V ISDN Card. I know that I have to use the ISDN4Linux so in modem.conf I set the I4l driver ,the msn and device. Modem.conf [interfaces] context=remote ;driver=aopen ; modem by AOpen driver=i4l ; isdn4linux - an

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

Hi everybody! I've managed to compile the bristuff patch on asterisk from Junghanns.net. I want to run this on the quadBRI card built into the PC. As I said, driver and asterisk are compiling well, but "asterisk -vvvvc" bombs with this message: [pbx_dundi.so]Nov 24 15:57:53 WARNING[1076754432]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined

Hi all, I'm having serious problem getting zaphfc to work on my box. I d/l'd bri-stuff-0.1.0RC3/RC4a and followed the instructions to the dot. Everything builds fine. But, when 'make load' the whole machine will freeze. Anyone had the same problem and managed to solve it? I'm using a Billion HFC PCI card on Trustix 2.0 running kernel 2.4.26. As a side note, I feel that

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem.... With the following line: exten => s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail("SIP/201-2f1e", "201") in new stack Sep 2 12:41:09

Hi, Bit of a puzzle this one - let's see if anybody else can shed some light... I Ghosted my Asterisk box to build one for a colleague. I added one HFC card to make the total two and amended zaptel.conf and zapata.conf accordingly. I tested it with my handsets on my ISDN lines and it worked AOK. I unplugged the box and took it to his house and plugged it in. He doesn't have any ISDN

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

Hi, I've been playing round with Asterisk on Redhat 9 (2.4 Kernel) and was experiencing bad echo problems using Budgetone 100's when calling analogue lines in uk (Isdn4Linux / Digi Datafire). Calls to other ISDN and mobile network seemed ok although not much testing done. I've tried installing Asterisk on a faster processor (P4 3.0 GHZ) with a 2.6.8-24 kernel to see if that helps.

Greetings to everyone! I am new to Asterisk and ISDN modules so i tried to follow many of the articles about capi, bristuff, mISDN and so on. Now I am working in mISDN but every way I try I have compiling errors! The HFC PCI card is a Digi Datafire Micro V. The bristuff give me the error "invalid module format zaphfc.ko" and I don't know how to compile and load it. The capi

I'm experimenting with asterisk. This is my environment: - Debian sarge (vanilla kernel 2.4.29) - Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g - Two sip phones (One cisco 7905 and one soft-phones X-Lite) - Digi International Datafire Micro V (Europe) (rev 02) (zaphfc) After two days of work now I can call between the two sip phones and I can call from sip phones to outside numbers via ISDN card.

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

I have a "Cologne Chip Designs GmbH ISDN network controller" in my asterisk box, and an ISDN line connected up to it. I downloaded the bri-stuff 0.1.0-RC4a package from junghanns.net and used that to compile asterisk using these instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install When I try to call my asterisk box, my sip phone rings twice and then the

Hi everybody, I have an euro-isdn bri line with it's own NT1. I connect to the NT1 my * box using a cologne chip pci card ( I run bri-stuff RC4a) and another pc ( a linux box used as proxy, firewall etc.). All seems to work fine, but when I'm surfing the internet during a telephone call ( in such situations both boxes are connected and both NT1 b-channels are busy) if someone

I have grouped two capi controllers using * as the outgoing msn. when the first two channels are busy normally and I try to use a third channel the channels from the second controller are used. But when a channel is occupied by the capisuite fax and we need a third channel asterisk responds with "every one is busy congested at this time". I think the same behavious would have happened

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load => res_parking.so load => chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -----BEGIN PGP SIGNATURE----- Version:

Hi there asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs; here some info: *CLI> realtime load sipfriends name 104 Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104' Jan 13 11:52:21 DEBUG[8928]: