similar to: Alarm events + asterisk dies

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

I don't know what the following means : [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 3, state 6 [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Polarity Reversal event occured - DEBUG 1: channel 3, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 294924 [Sep 14 10:41:25] DEBUG[4438] chan_dahdi.c: Polarity Reversal detected and now Hanging up on

Here's some freaky stuff coming from Areski CDR tool: 101. 2008-10-13 03:41:23 DAHDI/1... 0000000 "unknown" <0000000> BackGround silence/5 s ANSWERED 00:20 102. 2008-10-13 03:11:30 DAHDI/1... 0000000 "unknown" <0000000> BackGround silence/5 s ANSWERED 00:21 103. 2008-10-13 02:41:23 DAHDI/1... 0000000 "unknown" <0000000> BackGround

Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =

I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how

I've been a happy user of asterisk for over a year just for a small home setup (a Digium TDM400P with one POTS line and three internal extensions plus a couple of SIP phones). I recently moved from running Fedora Core 6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8, using the RPM package for * 1.4.18 and zaptel 1.4.9 . It worked fine for a week, but suddenly I can't

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

How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration? We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P. Twice today (first time in over a month) we received a Yellow Alarm on the TE110P. I have been able to clear it easily by restarting zaptel. Thanks in advance! -------------- next part -------------- An HTML attachment was

Guys.. I just saw this for the first time... I did some google and wiki without any luck.. what does a red or yellow alarm mean in zaptel? Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected alarm on channel 2: Red Alarm Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm cleared on channel 2 This just happened by itself..

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hello, I would really appreciate some help/guidance with this problem. First of all sorry for the long message. I would file a bug, but do not know if it is my fault, dm-cache, qemu or (probably) a combination of both. And i can imagine some of you have this setup up and running without problems (or maybe you think it works, just like i did, but it does not): PROBLEM LVM cache writeback

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

https://bugzilla.mindrot.org/show_bug.cgi?id=3188 Bug ID: 3188 Summary: Problems creating a second ecdsa-sk key for a second Yubikey Product: Portable OpenSSH Version: 8.3p1 Hardware: Other OS: Linux Status: NEW Severity: normal Priority: P5 Component: ssh-keygen

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

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

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I've done this, but I get: Sep

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

Hello I just installed a TE110P in a Dell Poweredge 750 (rackmount), which is connected via crossover T1 cable to a adit 600. Anyway, I've encountered an array of errors, which I believe I have narrowed down to the 22 IRQ misses I encounter on zttool. I've noticed that the te110p and the usb device share an IRQ. Could that be the source of my woes? Would disabling the usb device in the

Check with your telco. We had the same problem on 1 of our PRI's, every day at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly. Turns out the equipment at the CO was going into a test loop at that time because of a forgotten setting by a tech. Man, what a finger pointing exercise that was. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: