hi list i have an issue related to queues.conf i have configured the code like below extensions.conf [default] exten => 800,1,AgentLogin() exten => 666,1,Answer() exten => 666,2,Queue(hotline) #include "aheeva_dialplan.conf" ==============================queues.conf [hotline] member => Agent/1000 ==========================aheeva_dialplan.conf [agents] exten => 800,1,AgentLogin() exten => 666,1,Answer() exten => 666,2,Queue(hotline) ============================== agents.conf [general] persistentagents=yes [agents] musiconhold => none agent => 1000,,Agent agent => 1001,,Agent ======================================================================================================when i call the 800 i have call established and i can enter the agent ID 1000 without issue,the problem is when i call the 666 the call hangup please find the log -- Executing [h at agents:3] Hangup("SIP/1001-b7c8fc90", "") in new stack == Spawn extension (agents, h, 3) exited non-zero on 'SIP/1001-b7c8fc90' -- Executing [666 at agents:1] Answer("SIP/1001-b7c8fc90", "") in new stack [Oct 20 15:34:00] WARNING[19179]: pbx.c:1832 pbx_extension_helper: No application 'Queue' for extension (agents, 666, 2) == Spawn extension (agents, 666, 2) exited non-zero on 'SIP/1001-b7c8fc90' -- Executing [h at agents:1] GotoIf("SIP/1001-b7c8fc90", "0?3:2") in new stack -- Goto (agents,h,2) -- Executing [h at agents:2] AHEventsProxy("SIP/1001-b7c8fc90", "MSG_TYPE_TERMINATE_CALL::::1319124840") in new stack AHEventsProxy: Channel [SIP/1001-b7c8fc90]. Data [MSG_TYPE_TERMINATE_CALL::::1319124840] -- chan is SIP/1001-b7c8fc90 AHEventsProxy: Send To CtiServer: socket:[67]. message:[41,1319124840^^^^Ipbx01^~] -- Executing [h at agents:3] Hangup("SIP/1001-b7c8fc90", "") in new stack == Spawn extension (agents, h, 3) exited non-zero on 'SIP/1001-b7c8fc90' srvradio*CLI> could you please tell me what is wrong best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111020/f873630a/attachment.htm>
On Thu, Oct 20, 2011 at 11:01 AM, salaheddine elharit < salah.elharit200 at gmail.com> wrote:> [Oct 20 15:34:00] WARNING[19179]: pbx.c:1832 pbx_extension_helper: No > application 'Queue' for extension (agents, 666, 2) >This line here indicates to me that you don't have app_queue.so loaded. Try, from the asterisk cli, the following: module unload app_queue.so module load app_queue.so And report back any error messages that may pop up. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111020/f54e6a75/attachment.htm>
On Fri, Oct 21, 2011 at 1:15 PM, salaheddine elharit < salah.elharit200 at gmail.com> wrote:> hi > here is my extensions.conf and aheeva_diaplan.conf ><snip>> if you can see theses files and tell my if there is any wrong > > regards > > >These configuration files you sent me don't seem to match up with the dailplan CLI you showed earlier. Please, do the other things I asked about in my last email, and let's move forward from there. Also, let's keep the emails on the list. For reference, I've included my requests below: 2011/10/21 Warren Selby <wcselby at selbytech.com>> Please do the call again, this time please show us the output also with a > sip debug and a zap debug. > > These are both very old versions. The current release of asterisk is > currently five generations newer than what you're using, and Zaptel isn't > even used anymore, the tool was renamed to DAHDI. It may make more sense to > update to the latest version of at least the 1.4 branch of asterisk > (currently 1.4.42 I think?) and make the switch to DAHDI. This will require > some effort on your part, so don't do this without planning on a production > box. > > I don't know why you only need 3 numbers for your second provider, perhaps > that's all that they are sending you? You will probably need to ask the > provider why they are not sending you the full number like you're expecting. >-- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111025/45fd1822/attachment.htm>