similar to: queue members with multiple devices (bug 4759)

For whom asked me support for capi devices, that's here: http://www.junghanns.net/asterisk/ I'm using a AVM B1 card. also AVM passive card (FRITZ!PCI) works.... Then is you use SuSe all is configured by yast... Hello, probably is a feature what I'm asking for but because of my inexperience to asterisk this is my question: I've configured CAPI ISDN to receive calls. When I

Hello, I have two SAIS connected to one linux box. I want to monitorize the two sais, but the NUT only can monitorize ONE. I configure the port in auto in ups.conf but it only see one. [MGE1] driver = newhidups port = auto [MGE2] driver = newhidups port = auto MGE1 and MGE2 say the same values. The two instances exists, but they are "linked" with

I also have some other trouble. How the I send the caller to voicemail (next extension) if the Member => SIP/phone stops answering for a defined period of time. I cant figure out if this would work (from queues.conf): ; If you wish to remove callers from the queue if there are no agents present, then set ; this to yes. Note that this is for use with dynamic queue members! ; ; leavewhenempty

Hi. We have people with two or more sip phones. One wireless and one wired. So this is the case: Person A with two phones wants to have a queue for his incoming calls. So when he answers one of the two phones, the other phone should not ring. But when he isn't talking in any of the phones, they both should ring. Does that make any sense? This what I have for people with only one

Hi Everyone, I am doing a queue show and I can't see any column that shows a queue member priority. Is there any other command that can show the member priority on the Asterisk 1.4x CLI? We are using this format of dialplan to login agents: exten => 123,Answer() exten => 123,n,AddQueueMember(500|Local/${CALLERID(num)}@from-internal/n) exten => 123,Hangup() ^^^^ Where 500 is the

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond

Hi all, I'd like to know the best way to deal with queue member that are reached trough a SIP trunk. Let me explain: 1) "Master" asterisk box with my call queue 2) "Slave" asterisk box with a channel bank interface the two boxes are connected trough a SIP trunk, and the dialplan in the 1st box connect (for eg.) Local/0030 at gw to the 30rd channel on the 2nd box. All

Hi, Asterisk 1.4.31 here. I have a queue with members defined, and those member have member names "member 1", "member 2", etc. They are in a realtime DB. When I modify those member names (column membername) the changes aren't reflected in the queue status ("show queues" from cli. They aren't reflected when a new call comes in, or when I reload the

Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo Ann Boon), call is returned to queue and the cycle continues. If the same agent doesn't

Only to signal that if you want to compile app_rxfax and app_txfax with last cvsses of asterisk you have to modify the patched versions of the apps directory Makefile to define the symbol _GNU_SOURCE example, the line to compile app_rxfax from: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c to gcc -D_GNU_SOURCE -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c

> Asterisk Project Security Advisory - ASA-2007-011 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Multiple problems in SIP channel parser

> Asterisk Project Security Advisory - ASA-2007-011 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Multiple problems in SIP channel parser

need transcoding to a|ulaw. I am using it with no problems (have g729 licenses as well though). A bit off topic, I have found some extra configuration that is not really in the docs (or I could not find them): fullname=Your full name country=gr language=en city=City province=Province phone_home=+fullinternationalnumber phone_office=+fullinternationalnumber email=your at email.com

bonjour, Je voudrais faire une analyse des correspondances multiples avec R. avec les repr?sentation graphiques correspondantes avec R. je ne sais pas comment proc?der .. en vour remerciant par avance Faouzi

2011/6/14 Kevin P. Fleming <kpfleming at digium.com> > On 06/14/2011 03:11 PM, Olivier wrote: > >> Hi, >> >> 1. Is there any manual entry about modprobe's options relative to a >> given Dahdi driver (wctdm24xxp, for instance) ? >> >> 2. When loading a wctdm24xxp driver, is there any parameter to pass to >> modprobe to configure a span in