similar to: Manager API - standardization?

Is there any reason why IAXPeers output is different from SIPPeers output? The response has no Eventlist: start Ej. Response: Success Eventlist: start Message: Peer status list will follow Event: PeerEntry Channeltype: SIP ObjectName: 1001 ChanObjectType: peer IPaddress: 192.168.175.1 IPport: 63772 Dynamic: yes Natsupport: no VideoSupport: no TextSupport: no ACL: no Status:

There are 6 different ${QUEUESTATUS} variable values defined in asterisk 1.2, I am attempting to make sure I have a full understanting of when they would be set; If someone could correct errors with these definitions ot would be appreciated; TIMEOUT - the max time specified in the queue command elapsed, only checked between retries so may not be 100% accurate. FULL - the number of callers

Hello everyone. I got a little problem in here: I want to set up a queue so that if anything of these happens: a) No agents logged in b) All agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. <?php $strHost = "127.0.0.1"; $strUser = "cron"; $strSecret = "1234";

Hello, I'm trying to write a php script that issues the QueueStatus and Queues manager API commands to Asterisk and records the returned event data into a MySQL table. On voip-info.org, I see that QueueStatus returns such things as Queue, Max, Calls, Holdtime, Completed, Abandoned, ServiceLevel, and ServiceLevelPerf. However, there are no descriptions for these values. I've tried

Hello all, I'm trying to prevent answering a channel when a queue is either full or has no members. It seems I'm forced to answer a call before I call Queue() or else the audio is in the early media (which is unacceptable because of the short duration of early media on ISDN). Is there any way to let Queue() automatically answer a channel if the call is going to be placed in the

Hello all, I've a question regarding the values "completed" and "abandoned" that are returned by the manager command "queuestatus". What is the timeframe for these values, are they counted since the last asterisk boot, or per day, or is the timeframe configurable? Thanks and Regards Markus

Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas?

I am wondering how many and how often manager connections can be setup and torn down reasonably. here is the scenerio... I have 10 to 20 agents on two queues one with priority over the other I changed this the day before I also implemented a php program that runs every 8 seconds on an automatic refresh It establishes a connection to asterisk and runs a mysql query to update the database

Hi everyone, I'm having a trouble while developing monitoring tool for queues. I'm using C# Sharp & I follow the instruction on http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+C+Sharp. My question is how can I get information about how many people are there waiting on queue with this asterisk manager? I want to make that every time there is a call come into the

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>

Hey Everyone, Have a couple of questions here.. Scenario 1: We are working with a client that currently has one support queue with about 10 agents. They are starting to get pretty long hold times for their customers and they have requested three queues. Queue 1 will have a timeout of 4 minutes. After that it will move to Queue 2 which has a default timeout of 3 minutes. After that we will

I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Par?ina Lama Computers Split

Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

The Asterisk Development Team has announced the release of Asterisk 11.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.23.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi there It's possible configure realtime mysql in Asterisk with a non standard sippeers table? I need using a sippeers table from other system (non Asterisk). This table has a minimal configuration. Thank's Att, *Rafael dos Santos Saraiva* <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230> -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.