similar to: E1 PRI problem with TE205P

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

About every hour I see the yellow alarms on all or a number of channels of my PRI which is connected to the dutch telephony network, Asterisk keeps on working fine.... Here's an example where channel 1-24 went into alarm: WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 1: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event):

hi after some testing with A@H, i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 i load module zaptel and

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hello - I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone =

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

Hi ..., I've a TE205P card installed in my asterisk box.Port 1 of my card is connected to service provider.From port 2 I want to connect Avay Partner system. what type of cable I require to connect the partner system (straight/cross over). How the call routing from outside will be done to epbx. Thanks inadvance regards rudra -------------- next part -------------- An HTML attachment

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 am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm

I've been working with Digium, but hopefully someone on-list has an idea... The card will transmit audio to the T1, but everything it receives is completely unintelligible static (in place of received audio only, not all the time). I've verified everything on Digium's troubleshooting checklist and they were unable to find anything when they logged in. zttest yields 100% for the

Hi guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

I have a TE205P, jumpered for E1, added the missing wct4xxp-line to /etc/modprobe.d/zaptel, zaptel.conf is just span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 Which, according to my reading of the documentation I could find, should be correct. ztcfg doesn't complain about anything, yet all I ever get is RED

Hello, I am working on a project and have a few questions. I want to connect one port of the TE205p to the PSTN, and another port to the PRI port of a PBX. Basically Asterisk will sit between the PSTN and the existing PBX. Are there any gotcha's I need to be aware of? Will the existing PBX be able to dial through asterisk as if it was a PSTN connection? -- Isaac McDonald Got VoIP? imcdona

Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI with my Asterisk using a Digium TE220B card. The Panasonic is master clock and Asterisk is slave, additionally the Panasonic takes the clock from the PSTN E1. This scheme works and I?m able to make and receive calls the problem is that some times the Asterisk lose the synchronism with the Panasonic and the log shows this

Hi, I hope someone can help me with this.... Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2 System is located in Australia, so as technologies go, I believe it is twist on the euro standard for the E1 signalling. Here is the situation. The TDA E1 card is set in cross over mode and I am using a functional