similar to: queueing question

Hi all, I wanna setup a queue system that when a user presses a single button when queueing, he/she will quit to voicemail system. I've looked at different docs for a while and found the context in queue.conf would be useful. However, I've tried several times to use the context but without any success. Could anyone gives me some samples on how to use the context? Any reply would be much

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member

I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before

Hello again, I have noticed with Queues and roundrobin policy that if even if a timeout is set for a queue, Asterisk keeps ringing an available member of the queue after the timeout expires. This continues a few times before the next available agent is tried. I am using CVS of August 17 but I have read in the list that roundrobin worked fine since earlier in August. Does anyone know if this has

Hello all. I'm having trouble resolving an issue with our Asterisk system that seems to have popped up recently (no one knows for sure when the issue started). I'm still somewhat of a Asterisk newb and have been tasked with administrating the system as the previous administrator has left the company. Within asterisk, we have a Zap interface setup connected to a PRI that we have through a

Hello group, I tried a lot to enlarge the frequency (i.e. more announces, low wait between). according to config, every 30 seconds the announcement should take place. In fact, the first periodic announce is done after 2 minutes? What is my fault? Thank you Regards Paul # zypper if asterisk Loading repository data... Reading installed packages... Information for package asterisk:

I am working on this app to dial two external numbers. The second is dialed after the first hangs up. I have simplified things down to: exten => 3852,1,Dial(zap/g1/3964,10,g) exten => 3852,2,Wait(2) exten => 3852,3,Dial(zap/g1/7757,10,g) exten => 3852,4,Hangup Here is the debug: -- Accepting call from '0000000000' to '3852' on channel 0/23, span 1 --

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

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]

Hi, I have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current) Today when I setup queues for the first time (with one member in my default queue), I got some really strange behaviour, aside from my hysterical laughing after hearing the default MOH =) I only have one SIP hardphone I'm testing with right now, so I tested using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog

Howdy, Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel, zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source, asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source, Sangoma A104d with firmware that is probably a year old). I plugged in an RBS T1, ESF, B8ZS, wink start, and MF signalling. I stuck with the defaults that the wanpipe build

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second

I have this in extensions.conf: exten => 1,1,Dial,Zap/g9/4439568899/|24 exten => 1,2,VoiceMail(u8004) and this happens: -- Playing 'js-joe-trvm' -- Executing Dial("Zap/2-1", "Zap/g9/4439568899/|24") in new stack -- Called g9/4439568899/ -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Nobody picked up in 24000 ms -- Hungup

SBC installed a T1 ADTSe (Digital Trunking Service Enhanced) e&m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now

Skipped content of type multipart/alternative-------------- next part -------------- Console logs from Asterisk A: Executing Dial("SIP/test0-5821", "Zap/6/327557670||Tt") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 6/327557670 -- Zap/6-1 is proceeding passing it to SIP/test0-5821 -- Accepting UNAUTHENTICATED call from 195.66.73.122:

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Here's my queue.conf: [general] persistentmembers = yes [738] musiconhold = empty ;musiconhold = default ;announce = q-738 ;strategy = ringall strategy = rrmemory