similar to: TDM400P off-hook detection problem

I have a running * with a TDM40B board in it. I have 3 analog phones that works (rings) perfectly when connected to a Telco POTS line. When connected to the Digium TDM40B (with FXS port), I have problems with 'ringing': 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri.... ring... ri...) and the 3rd

I have a serious problem that repeats very often, after 30 - 50 calls and I can only solve it by stopped and restarting * :-( After a while, * seems to loose track of something. When an ISDN call from PBX needs to go to the Telco, I get 'Ring requested on unconfigured channel 255/255 span 2' It's always channel 255/255 ??, the 'span' number is random Setup: * between Local

Hi, I'am new to * and googled/read a lot, but did not find (yet) a lot of info to do the above. Some months ago, I did find a 'story' from somebody having put * between his PRI and current PBX as IVR, but I can not find it back :-( Any help/pointers are appriciated. Txs alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs

Hi, I have just started to setup and configure my asterisk box and am having trouble with it. I have a dlink nic in the box as well as the digium card. When the nic is in there by itself it works, but when I put the tdm400p in, there seems to be some sort of conflict with the network card. The tdm appears as another type of network card, which I don't think it should, should it? It shows

Hi all, How can I make a X100P ZAP channel not answering to any incoming calls? Thanks. Newbie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/73c47a3f/attachment.htm

Hi, I am new to Asterisk and have some general questions _before_ I start buying equipent to install and get everything up-and-running. (this means I have no running Asterisk (yet)). I have read already a lot of doc, but some things are not clear to me, since I'am inexperienced in Asterisk and PBX. The goal: Make PBX using Asterisk and Cisco 79xx equipment for 25 phones (1 x 7970, 4 x 7960,

Hi, I wanted to configure the following : 1. VPN + some other special connections (TCP 82,8282,23,22 and ICMP) to have priority over the rest 2. special upstream for our updating system on port 4000 3. within the VPN tunnels citrix traffic ( TCP 1494, 2598) + icmp has priority I need this for both incoming and outgoing traffic as it is the bandwith managment config on a central system from

Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine

Hi, We are using kernel 2.4.24, tcng version 10b. I''m trying to do some policing in the ingress queue of the internet device. Until now we had configured some filters dividing traffic into queues, and on these queues the Double Leaky bucket meter was applied. The idea is to have a minimum of bandwidth assigned per class (the cir values) and a maximum (pir values), just as with the

Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

On 10/1/2020 8:28 AM, Rowland penny via samba wrote: > On 01/10/2020 13:17, Jason Keltz via samba wrote: >> So why is it that winbind renews the ticket on the original system, >> but on the system that I ssh to, it does not. > > Do you have 'winbind refresh tickets = yes' set on all the systems ? Absolutely.? In fact,? both systems are using the identical smb.conf,

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

On 10/1/2020 8:34 AM, Rowland penny via samba wrote: > On 01/10/2020 13:30, Jason Keltz via samba wrote: >> On 10/1/2020 8:28 AM, Rowland penny via samba wrote: >> >>> On 01/10/2020 13:17, Jason Keltz via samba wrote: >>>> So why is it that winbind renews the ticket on the original system, >>>> but on the system that I ssh to, it does not.

I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I'm running rsync 2.5.1 on solaris 7 in server mode and getting the above message when entering the following command: rsync -vraz --delete hostname::test ./ this is my rsyncd.conf use chroot = no max connections = 25 syslog facility = local 3 read only = false log file = /var/log/rsyncd.log pid file = /var/log/rsyncd.pid lock file = /var/log/rsyncd.lock [test] path = /export/home/mrl315

Hello, Is it possible to share a single host's Logical Volume among multiple local KVM guests which mount it read-only? I'm asking this because I have an idea to run multiple idential KVM guests (they all have exactly the same software installed on them), booting them from a shared local Logical Volume read-only root file system, or alternatively let them share the bulk of the software

Vinay S. Belgaumkar wrote: > Ok, I am trying to configure llvm to compile benchmarks as suggested by > John, and after running configure, I get the following error while > running make in the /test/Programs/External/SPEC Hmmm. It sounds like the Makefiles are not creating the gccld command line correctly. Some questions: 1) Can you send us your Makefile.config? 2) Can you tell us

I am having problems getting a Nicherons TDM400P wildcard clone to dial out. Everything appears to be configured correctly, but although I see call progress, it never seems to actually pick up the phone. (The following is a test of 911 emergency, where I substitute 811 [repair service] as the actual number dialed.) *CLI> -- Executing [911 at from-internal:1]