similar to: Strange Segmentation fault

I have a multi-port PCI-X card that is causing me some grief. An older version of the system bios allowed this to work correctly, and the on-board devices to work correctly. the newer system bios has fixed a 'bug' where the interrupts were routed incorrectly according to the PCI bridge specification. The system vendor indicates that I am likely missing the support for multiple IOxAPIC,

Was there any resolution on this I also have a TE410P in an box with an Intel E7501 chipset? -Vickers -------- Original Message -------- Subject: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server Date: Tue, 8 Feb 2005 11:13:24 +1030 From: Peter Childs <peter.childs@nec.com.au> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750

-----Original Message-----From: John R Pierce <pierce at hogranch.com> Reply-to: CentOS mailing list <centos at centos.org> To: centos at centos.org Subject: Re: [CentOS] SCSI drives and Centos 7 Date: Sat, 29 Apr 2017 11:19:38 -0700 On 4/29/2017 10:49 AM, Gregory P. Ennis wrote: > I am remote from the unit today, and do not have a good way to look at > the board today, but the

Hello. I have been going thru the wiki and asterisk related sites and have not been able to find any documentation about how to configure an Adtran TA750 channel bank. The remote disconnect supervision doesn't seem to be working, when the remote caller hangs up asterisk takes up to 30-45 seconds to hangup the call. Can somebody help? Thank's

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. " WARNING[8446]: chan_oh323.c:3555

-----Original Message-----From: John R Pierce <pierce at hogranch.com> Reply-to: CentOS mailing list <centos at centos.org> To: centos at centos.org Subject: Re: [CentOS] SCSI drives and Centos 7 Date: Sat, 29 Apr 2017 13:31:11 -0700 On 4/29/2017 12:42 PM, Gregory P. Ennis wrote: > what does `lspci` have to say about this raid card ? > > John, > > Thanks for the prompt

Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with

Hello folks. We are trying to interconnect an asterisk installation with a gnugk 2.0.5 installation to become able to use some H323 hardware that needs a gatekeeper (particulary an Ericsson WebSwitch 100). We have managed asterisk to dial H323 endpoints successfully (although calls are interrupted immediately after connection with "spawn extension exited non-zero"), but we could not

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model

Hi, I'm using asterisk with chan_h323 together with gnugk. chan_h323 and gnugk were recently compiled with pwlib-1.5.2 and openh323-1.12.2 as advised. When connecting asterisk directly by ohphone (without gatekeeper), everthing is fine. When using gnugk for usage control in routed mode, I find a funny situation in asterisk's H.323 debug: == New H.323 Connection created. --

I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323 channel driver. I have a Gatekeeper that gets H.323 calls from a Cisco GW. To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom 100, etc. Now i want send the numbers 083xxx into Asterisk. Easy, i'll just enter something like this into oh323.conf: gwprefix=083 And all my calls starting with 083