similar to: PRI error cause hangup calls

Hi, we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that are restarting after 5 - 15 minutes, before and after that we can make calls in and out w/o problems. The cards have been tested in two computers (Atholon XP 2200+ and Celeron 2.6Ghz), are on there own IRQ, not showing any IRQ misses in zttool. It's all correctly configuered - zaptel and zapata (attached). We

Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also

Hi everyone, I have a PRI conection on an * system running Asterisk 1.0.9, libpri 1.0.9 and zaptel 1.0.9.2 connected to an AXE 10 (APZ 21220 System 64) in the network side. I know the system and the wildcard I´m using are ok because I´ve used them before with other PRI connections (to a Siemens EWSD) without any problem. First the PRI didn´t work (I got the TE110P alarmed with red). After

Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf loadzone=dk defaultzone=dk span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 == ==

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I'm getting the following error at random intervals on my TE410P with Asterisk CVS-03/30/04-11:49:01-CEST. I have two spans active, one connected to my Telco, the other to a Siemens PABX. Both spans display this behavior at random intervals. All calls are dropped when this happens. Spans are not necessarily in use when this happens.

Hi, I have an IBM server running latest asterisk 1.4.x connected to a Siemens hi-path user a TE120P single-span. Approx every 8 hours (although not every 8 hours and sometimes 2 in a row) at exactly the same time I see the following errors Does anyone have any suggestions / ideas ? Thanks Ruairi Sep 24 11:56:30 asterisk01 -- MARK -- Sep 24 11:59:02 asterisk01 ntpd[3615]: synchronized to

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 all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

Hello my friends, I would like to understand the output from "pri intense debug span X", the Telco always says that their side is OK, but i always receive alarms, loosing connection, take a look: [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1: Recovering [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Unable to disable echo cancellation on channel 1 [...] [Sep 16

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

hello everyone, Last week I installed asterisk 1.2.24 with digium TE220B card. I have a problem with our PRI and Asterisk: the call be interrupted.It happens either PSTN-to-SIP or SIP-to-SIP,almost every call. After spending several days searching on internet, I found a lot of discussion about this issue and I have tried many, but it still.I am totally new to Asterisk environment and suspect I

Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a "straight" cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same.

Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. The "flaw" in the messages is the "Alarm cleared" message - The alarm cannot possibly be cleared because there is no physical media connected into that port!!! (BTW - All

Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P (same problem with various previous versions; same problem with different TE120P cards). The customer has a partial (10 B-Channel) PRI that when it is busy (eight or more B channels in use), tends to fail as shown below... [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown error 500 [Jan 26 23:00:31]

Today is the first time I've had any show stopping errors with asterisk since I started using it over 8 months ago... While I wait on the telco to check the circuit I thought I would post this here. Not sure yet but I thinks my 4-port PRI card is gone!!! I made a call and right when the person answered - BOOM. This is only a little better than it not working at all.

So it was working for several days without error then tonight while I'm watching users login to the modem pool - BOOM! The sync changes when it happens (which makes sense). Zaptel: Master changed to Tor2/0/2 Zaptel: Master changed to Tor2/0/1 Zaptel: Master changed to Tor2/0/2 Zaptel: Master changed to Tor2/0/1 Interrupts should not be a factor THIS time: [root@phone asterisk]# cat

"RED: Loss of signal (LOS): The equipment shall assume "loss of signal" when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their

"RED: Loss of signal (LOS): The equipment shall assume "loss of signal" when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their

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*