similar to: Asterisk queue not play muscinhold or hangup

Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1) [fromiax] exten => 9999,1,MeetMe(9999,qdxAa) exten => 9999,2,MeetMe(9999,qdx) exten =>

I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the "s" extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The

Hello All, I currently have an Asterisk Box, running a callcenter with 04 queues. I set queues.conf with "persistentmembers=yes" in the general section as follows: [general] monitor-type = MixMonitor persistentmembers = yes However when I perform any kind of restart in the Asterisk application, all agents are considered unavailable after that. Though when performing

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.

Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective

I'm stuck on queues! The way I read what documentation I have found, if I set up a queue like this: [general] persistentmembers = yes [testq] musiconhold=default strategy = ringall timeout = 10 retry = 5 context = testing member => SIP/100 and then add into extensions something like this: [incomingiax] exten => 1234,1,Dial(SIP/100,10) exten => 1234,2,Queue(testq|tTH|||300)

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

Hello Asterisk Users, I guess the subject says the most of it; here goes some more detail: - Running Asterisk 1.2.14 - Objective: record all calls managed by a specific queue - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID} Facts: - If the UNIQUEID chan var is used in the MONITOR_FILENAME, before calling the Queue() application, the two legs of the call are not

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

Hi, I have a question about the definition of agents. The agents.conf file looks like this: [general] persistentagents=yes [agents] maxlogintries=5 ackcall=no wrapuptime=500 musiconhold => default group = 1 agent => 1311,1311,Tom agent => 1531,1531,Tim and here is the queues.conf: [general] persistentmembers = yes [queue1] musiconhold = default strategy = rrmemory servicelevel = 60

Hi , I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101 ) in a queue..When a caller arrives in queue , it lands on first 100 , 100 then does a blind transfer to 101 .. so that the caller can converse with 101 .. strangely enough the queue_log shows : 1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123

Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected ("estimated hold time is less than 2 minutes ..."). Now the caller gets an announcement of their sequence in the queue

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

Hi, My queue members use Local channels and their queue state is "In use" while their hint value is "Idle". Since I have Ringinuse=no, I'm experiencing issues such as incoming calls waiting too much because the agent's phone isn't ringing even though it's idle/free. I read somewhere that this is a "known bug" in 1.4 and should be fixed in 1.6. I

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122 at from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a

The Asterisk Development Team has announced the release of Asterisk 11.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this