similar to: Agent Dump

If I have in my dialplan [AgentQ] exten => _XX.,1,Dial(Sip/{$exten},120,g) exten => _XX.,2,NoOP(here we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus:

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

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 =>

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it

When recording inbound agent calls, if the queues use agent members (Agent/6000), we can get the calls recorded as agent-xxxx.yyyy.zzzz.gsm where xxxx is the agent number. However, if the queues use phone members (SIP/6000), the recorded filename is simply yyyy.zzzz.gsm. Is there any way of making the recorded file either agent-xxxx or even sip-xxxx where xxxx is the extension number. I had

I was wanting to use the new MuxMon application to record calls - it seems to be a "nicer" way of recording than the Monitor application. However, there is a slight issue with agents - we use the recordcalls option in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application. Is there any way to get chan_agent to

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

Folks, please, take a look at this asterisk log message: [Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file 'atcert' is unavailable, continuing anyway... [Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002 hungup on the customer. but: -bash-3.1$ whoami asterisk -bash-3.1$ ls -ls $HOME/sounds/atcert* 12 -rw-r--r-- 1 asterisk asterisk 10956 Oct 2 07:00

Guys. What does this error mean? -- Playing '/var/spool/asterisk/voicemail/intruder/201/unavail' (language 'sp') -- Playing 'vm-intro' (language 'sp') -- Playing 'beep' (language 'sp') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/intruder/201/INBOX/msg0000 format: wav, 0x812b4f0 -- User ended

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257 #3 0x08092b3f in handle_reload (fd=33,

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Phone - TDM430P - home* - IAX2 - office* - PRI - Telco I dial a busy number from the Phone. Home* shows this in the CLI: -- Executing Macro("Zap/1-1", "dial-wu|2922004") in new stack -- Executing Dial("Zap/1-1", "IAX2/andrew@wu-ast/2922004||g") in new stack -- Called andrew@wu-ast/2922004 -- Call accepted by wu-ast (format gsm) --

Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free ()

hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35

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

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

Hi, I'm playing with the agent/queue system. Everything work well with v1.0.3. but I want the 'Action: Agents' in the manager API that is only on the CVS version. So i switched to, but now the Queue/Agent system barely work. (my agent don't get the call) Where I can get a 'stable' CVS version? Or maybe, how I can solve my Queue/Problem? here is the detail: 1. I can