similar to: Zaptel slow dial out - TDM400P

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

All, I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all* of my incoming calls are coming up as FAXes. I had to disable my fax extension because every call to my POTS line was getting redirected to my FAX machine. After removing the FAX extension, if I call my POTS line from my cell phone, I get the following: *CLI> -- Starting simple switch on 'Zap/1-1'

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hi, I am having some trouble getting a TDM400P working, and would be very appriciative of some ideas. I have installed a TDM400P, and downloaded the appropriate files from CVS, compiled, installed and modprobe'd the devices.

Hi List, I have a test asterisk box with a TDM400P with 4 FXO modules plugged in. Yesterday I could use the box without any issues - no problems. This morning, the sound on the box was absolutely horrible. After some fiddling about, I have rebooted the box, and now asterisk refuses to start! Here's the message I get: Jun 9 10:45:53 WARNING[3297]: chan_zap.c:769 zt_open: Unable to

Howdy, 1 x TDM400P card with 1 x fxo module. 1 x BT Pots line. Location - UK Calls work fine outbound but i'm unable to pickup the inbound calls. Asterisk debug: Asterisk -vvvvvvvvvvcg *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack

Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel

Hi all, I'm trying to install a TDM400P card, and I need some help. Please, see below... after dmesg command: [root@voip root]# dmesg via82cxxx: board #1 at 0xD800, IRQ 5 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 00:09.0 PCI: Sharing IRQ 3 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I just received two TDM400P cards, but I'm having problems with them. The full info is at: http://pastebin.com/824079 Extra, I'm using : libpri-1.2.3 zaptel-1.2.10 On a x86 stable Gentoo box. Kernel: 2.6.17 gcc-4.1.1, glibc-2.4-r4 Is that an hardware problem? Should I try the other card? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog:

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24

Hello, I thought that my Digium TDM400P would be the right hardware to support the zaptel timer, and put the following IAX.CONF entry to test, (trunk=yes) in the example below [VHAX] type=peer auth=md5 username=whoknows jitterbuffer=yes ;trunk=yes secret=terriblesecret host=4.5.6.7 qualify=1200 disallow=all allow=ulaw allow=gsm ;allow=g711u ;allow=g711a But, it didn't work. So I had to

Calling out on PSTN over a TDM400P seems to generate a Red Alarm - whatever that is. I have another extension on the PSTN, and I can dial out over that. zttool shows no alarms. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db

Hello, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

Hi , I have an asterix (1.2.7.1) box with 3 TDM400P and 10 FXO modules, and zaptel (1.2.5). The problem is that 6 modules can't be initaized and i get the folowing error: May 7 09:17:37 WARNING[7739] chan_zap.c: Unable to specify channel 9: No such device May 7 09:17:37 ERROR[7739] chan_zap.c: Unable to open channel 9: No such device here = 0, tmp->channel = 9, channel = 9 ztcfg output

Hi to all, I have been using Wildcard TDM400P with four fxs modules and it was working fine now I have added another Wildcard TDM400P with four fxs modules . So there are total 8 ports for 8 hard phones. I have modified following configurations In /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1-8 in /etc/asterisk/zapata.conf context=headoffice signalling = fxo_ks

I have been given a 'code' file and a 'data' file which is designed for use in R. I have successfully installed the program onto my pc, but am struggling to actaully use the code. How do i prompt R to read the code file and run? Any ideas? Many thanks Mark ____________ Mark Edwards mark at miic.org.uk ******************************************** DISCLAIMER: The