similar to: Failed to read gains: Invalid argument

I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P card in E1 mode. I've recently noticed in my logs the following Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing '/etc/asterisk/zapata.conf': Jan 5 01:27:11

We have an asterisk box with the following configuration: - AMD Athlon XP 2400+ - 512 MB RAM - SUSE Linux 10.1 - a Digium card TDM400P with 3 FXO - another Digium card TDM400P with 4 FXS - asterisk 1.2.7.1 - zaptel 1.2.4 I already checked that those cards aren't sharing interrupts (by cat /proc/interrupts): 0: 14119786 XT-PIC timer 1: 10 XT-PIC i8042 2:

Sorry I should have stated that I've tried +x, -x, x.y and x and I still get the same. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: 05 January 2007 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

Hi, I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT,

Hi All... Well, I'm still experiencing LOTS of dropped calls since installing the new (non pri) T1 here... I keep noticing a few things in the logs when this happens, namely the "Wink/Flash" statements and the "Didn't get a frame" messages... Anyone got any ideas on if this is a telco issue, a wiring issue, or an asterisk issue? Been trying to track this down via all 3

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, When startin * it tells me: ================================ [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, PRI Signalling signalling Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf:

Hi There. I have the following setup : Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24 My problem is as follows : If I set up a very simple extensions.conf. when I dial from a fax machine, it seems as if no fax is being recognised. If I answer the call, I can hear the fax machine beeping. extensions.conf :

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hello, I'm trying to set the 3rd span of a new digium quad card as a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make

Using tdm400. While transfering a call from outside to another extensions, while this "outside call" is waiting with music, the "another extension" call hangs up suddenly, and the call is back to the "outside call" suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850]

Hi! I am trying to get my ISDN phone to work with my asterisk box. Now my asterisk won't start. Current situation: I have a cable from my Billion ISDN (Bipac V1.0) to my old NT1. The cable is crossed like this: 1 2 3 -> 4 4 -> 3 5 -> 6 6 -> 5 7 8 Then I have a cable from the NT1 to the ISDNphone(not crossed cable). Both cables are connected in the ISDN

I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

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

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

About once a week or so my Asterisk box stops responding to all phones. I can pull up the console, do whatever I want at the CLI but the only way to get things working again is to restart Asterisk altogether. I finally cranked verbose & debugging way up (and watched my log files go from 1mb/day to 100mb/day), but below I believe contains my problem. The next line is 1.5 minutes later where I

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.