similar to: T100P: Can't Make/Receive Zap Calls (Long Newbie Blah)

Hi All, Channels 25-28 on a customers PBX are regular Zaptel FXO cards that are hooked into 4 incomming phone lines. They are all in a group to do automatic rollover for outgoing calls (if channel 25 is being used, dial on channel 26, etc.). Sometimes when a user is dialing a number, instead of bridging to one of the FXO cards it goes and rings to Zap/1-1. This doesnt occur all the time but some

Hi, i was using rsync 2.4.5 for AIX 4.3 binary downloaded from http://rsync.samba.org/ftp/rsync/binaries/AIX/ . I now compiled the latest 2.5.6 on AIX 4.3.3 and created a tarball to use like the one available from http://rsync.samba.org/ftp/rsync/binaries/AIX/ . root@b50srv1:/usr/local/src# uname -a AIX b50srv1 3 4 000B14FD4C00 root@b50srv1:/usr/local/src# oslevel 4.3.3.0

Hello all, I just got the latest cvs snapshot and I'm trying to compile asterisk.. When I run nefeli:/usr/src/asterisk/zaptel# make clean; make linux26 Compile stops with errors... /usr/src/asterisk/zaptel/zaptel.c:6509: warning: passing argument 2 of 'class_device_create' makes pointer from integer without a cast /usr/src/asterisk/zaptel/zaptel.c:6509: warning: passing argument 3

Thank you Andy! I did''t find why I could''t use the fw match in ingress. Now it works. Now I ran into another problem: When I try to add a pfifo_fast leaf qdisc dor htb classes it seg.-faults. Is pfifo_fast a special qdisc? I can succesfully add other type of qdiscs. I found that a similar problem was reported in february on a MIPS architecture. root@zefir:/usr/src# tc qdisc

Hi, I configured an asterisk box with 1 Digium Wildcard TE110P T1/E1 Card 0 I setup the jumper in e1 position. my zaptel.conf : defaultzone=it loadzone=it #gestione PRI span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 # Asterisk starts correctly, I see th 30 channels. Anyway I cannot put

Hello, I have a problem with zaptel. I hope you can help me. I installed and configure zaptel. ZAPTEL.CONF span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = de defaultzone=de But the output of cat /proc/zaptel/* Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/CCS/CRC4 RECOVERING 1 WCT1/0/1 Clear (In use) RED

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Asterisk 1.0.9, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> .... [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 16 20:36:51 ERROR[6750]: chan_zap.c:6246 mkintf: Unable to get parameters Sep 16 20:36:51 ERROR[6750]:

Hi everybody, I've connected Asterisk 1.2.5 (libpri 1.2.2, zaptel 1.2.4, Linux 2.6.13.2) to an Avaya-Tenovis PBX via a PRI/E1-line. Calls from SIP-phones via * to the PBX work fine. However, incoming calls to * only result in: -- XXX Missing handling for mandatory IE 24 (cs0, Channel Identification) XXX -- which seems to be an * problem, because a Windows-fax-machine works fine on a PRI

Hi, I have a working Asterisk 1.2 server on kernel 2.6.22 with the OSLEC echo canceller on a Digium PRI card. I recently switched to kernel 2.6.23 with the MG2 echo canceller (nothing else changed). Each time I try to establish a call on the PRI line I get a congestion signal. in /var/log/asterisk/full: Feb 20 08:09:43 VERBOSE[10657] logger.c: -- Executing

Hi, I'm having trouble configuring and E1 link , I know the E1 is a PRI and the switchtype is 5ess. The problem is that everytime I try to dial I got and error saying that it was unable to open the zap channel . Devices are created in the /dev/zap directory and I can open them with a cat /dev/zap/1. I can also see the channels in /proc/zaptel/1. Thanks in advance [zaptel.conf]

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

Hi!I've a problem with asterisk,it doesn't start..it's stopped..I used the amportal start command and this is the result: SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk.

Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log

> > Hi, Remove usecallingpres=yes busydetect=yes from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues. Regards, Vidura Senadeera. Message: 3 > Date: Mon, 24 Sep 2007 12:29:40 +0100 > From: "Richard Young" <Richard.Young at intrintech.com> > Subject: [asterisk-users] Asterisk Dropping Calls > To:

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

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

I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI> module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an

Hi all, I?ve ran into a problem regarding D-channels. I have setup Asterisk (CVS-HEAD-03/21/05-16:41:57) on RH Fedora Core 3(2.6.10-1.770_FC3) I also have a digium wcte11xp card for connetivity to the PSTN(E1). When I start zttool i see that Current Alarms changes between Recovering and Blue Alarm/Recovering. I started to see these problems after i moved the digiumcard from one PCI slot to

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm