similar to: Strange FXS disconnection problem.

Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the

HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and

Hi, I'm trying to get asterisk to auto-dail out. I created a *.call file with the the top of it being "Channel: Zap/1/2609944", which should have connected to Zap channel 1 and dial out to 2609944, but It did not do so, asterisk would say a call was completed to Zap/1/2609944 but I never heard that phone ring. So I tried just putting "Channel: Zap/1" at the top of

Please help me I have been tired now... I have Counter strike source installer for the windows 7.. and it is running properly on the windows.... In that installer I Have total 3 files and 2 folders. FILES 1. CSS_Full_game 2.CSS_PATCH 3.CSS_MASTERSERVER_PATCH FOLDERS 1.CSTRIKE 2.HL2 I dont have any DVD for it.. and i have to install that counter strike on the ubuntu.. i dont want to purchase

Hi All: We have some patches to port klibc to ArmV8. The first set of patches from Neil Williams fixes the build issues. The second set of patches from Anil Singhar fixes functionalities. The following files have changed in the second set of patches ------------------------------------------------------------------------------------------ usr/include/arch/aarch64/klibc/archconfig.h

Hi List, I'm having a bit of an odd problem with asterisk and outgoing zap calls. Tzafrir has been kind enough to help me get the logging sorted out so I have some idea of what's going wrong, but I'm a little flummoxed. Essentially the symptoms are as follows; Make a SIP call from Cisco 7960 or 7940 to asterisk, where it is routed out on a ZAP (x100p) line. After

Hi all, I was wondering if anybody has experienced a fixed delay about 30-40seconds while logging onto the samba domain running samba version 3.3 from Windows Vista Business Cheers, Abey

I know the recommendation it to not run more than 2 T/E100P's in a system but what about X100P's.. Usually there are 5 PCI slots in a system, has anyone tried 5 x X100P's in a system? Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I'm testing Teliax tall free number line and I'm experiencing long delay about 1sec. during conversation. When I call myself over FWD the response is normal no delay or cut messages. When I call my number over FWD the is a long delay, welcome message usually cuts off few first words and during conversation my voice arrives after about 1sec. delay. Since, the 800-number is only accessible

DTraceToolkit has a dtrace script (Proc/pidpersec.d) that refers to a variable mpid as below profile:::tick-1sec { printf("%-22Y %8d %6d\n", walltimestamp, `mpid, pids); pids = 0; } Where is `mpid getting picked from (it isn''t declared anywhere else)? What does the character ` in `mpid signify? I do not see any reference to this variable in the dtrace

Hello, I''m using tc to limit the bandwidth of our wireless customers. I have a working script, but I''m not happy with it. I''m trying to write a more sophisticated script, but when I run it, it give me this error: RTNETLINK answers: File exists I have no idea what this error means or how to fix it. Here is a portion of the script (the whole script shapes several

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hi, I am not sure if this problem has been discussed already. I am running red-hat LINUX 2.2.14 and the version of iproute that comes with it. The problem is that whatever I give as a command it tells me RTNETLINK answers : Invalid argument Example : tc qdisc add dev eth2 handle 10: root estimator 1sec 8sec prio bands 3 priomap 0 1 2 I have tried all kinds of combinations and it refuses to

hello, a friend of mine have this configuration: 10 x PC -- router/linux/rh8 -- ADSL Modem -- ISP let''s say that the bandwidth is: 5M and 800K he does dc++ and counter-strike, so let''s say the UP is full, and the ping from the counter server is 300ms, the server cut the connection, and no more game, the player is unhappy. The normal ping is 50ms. so he thinks to put some

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

Hi folks, this is specifically directed to Australia Asterisk users...... We are having a roblem with x100p 's when dialing STD. Upon receipt of the approximately the 5th (out of the ten) PIP's asterisk will hang up.... Now I am wondering if others are suffering the same problem ?? Any ideas ?? (it might exist on other cards, but so far I have only noticed the problem on x100p's).

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to

Dear All ~ I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN & try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the