similar to: ringing... or lack thereof

I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? joe a.

I've recently been trying to track down the root cause of my server's persistent issue of thrashing horribly after being left inactive. It seems that the issue is likely my nightly backup schedule (using rsync) which traverses my entire 50GB home directory. I was surprised to find that rsync does not use fadvise to notify the kernel of its use-once data usage pattern. It looks like a

I am TRYING to use the most recent release of wine. Am I correct to assume that there is no longer a .wine/config file? Am I also correct to assume that all documentation still refers to the .wine/config file, and gives NO F*CKING help about how to use the new system? If these two assumptions are correct, it seems to be a real stupid way to do things. How is a new user EVER supposed to

Sorry if this is a daft question but when a PSTN call comes in on my X100P the console shows the following; NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)...

On 13 March 2015 at 14:04, Matt Hamilton <efes9999 at hotmail.com> wrote: > We use the ringall strategy for a small queue with 4 members. When a call > comes in, if one of the members is busy, all the phones except the busy > phone rings (as intended). While the other phones are ringing, if this busy > phone becomes available again, we would like to have it start ringing. >

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone

I'm using a modified "default config" file for extensions.conf, the one that uses macro-stdexten to handle the stations. We use a TDM30 card for our stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the "other" instrument ringing. But once the ring timeout has expired, the call then drops into the *original

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle. Is this possible? I played with ringinuse (queues.conf) and callcounter

All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Ray Burkholder > Sent: Monday, January 19, 2004 7:38 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] ADSI phone vs. IP phone (and > proper implementation thereof) > > [...] > I'm wondering if

Hi, For some time now I have noticed that our RBS T1 (asterisk 1.4.35, Dahdi 2.3.0+2.3.0, TE410P) often has channels stuck in the state "Ringing", like this poor chap who got stuck on two calls in a row, apparently: [excerpt from "core show channels"] SIP/7157997-0000534b 7760308 at business:1 Ring Dial(Dahdi/g0/7760308) DAHDI/3-1 5130262 at from-pstn:1

I am having some trouble creating a hierarchy of virtual classes (akin to the class structure in the 'Matrix' package). I think they arise from my not understanding the best way to specify virtual subclasses of a virtual class. please see questions below code. setClass("mom") setClass("kid1", representation("mom", "VIRTUAL"))

Hi I am using DISA to get my Polycom SoundPoint400 with H323 firmware to connect to * I have it working, but when I dial SIP end points there is no ringing tone on the phone. DISA gives dial tone but does not give ringing (if I understand correctly it is because it expects to transmit sound created by terminating side of the call) Is there a way to make DISA application to generate ringing

I'm using a SNOM 360 with Ver 4.3 software. Asterisk is.... Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff + Head) I've used the wiki info to set up some lines to monitor some internal extensions. When the extension is rung - the lamp comes on, when the call is answered, the lamp goes off.. I was expecting something a little more exciting - like the lamp to flash when the

Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 -> 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 -> 10 and 11 -> 15 are busy) Within the range of Agent 1 till Agent

I have a problem with one way audio on Sip and I guess it may be a NAT issue, in the example below 204 is rung by 208 (xlite external) I dial perfectly but when I get to the answering of the Asterisk, I can hear audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring the voice mail , Asterisk answers and then cannot hear my password... I have put the Ports Forward

I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'. Everything appears normal, but the status of the members never changes from 'not in use', even if they are being rang or are in a call. Members are added like so: queue add member SIP/1406 to marketing penalty 0 as SIP/1406 state_interface SIP/1406 And they are present as a hint: exten =>