Confirmation not necessary. I just found that this bug has already been reported:
http://bugs.digium.com/view.php?id=3861 -----Opprinnelig melding----- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P� vegne av Bjorn Sendt: 4. juni 2005 19:00 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [Asterisk-Users] Possibly a bug? (Was: Setting up calls through themanager interface) Thanks Tony/Olle, you both answered my question. However, now I am wondering if there might be a bug with Asterisk, or if I am simply misconfiguring something. This is the scenario: Taken from extensions.conf (a little simplified): Exten => 1234,1,Queue(managerqueue) Exten => 1234,2,Hangup Then, telnet'ing to asterisk manager interface and issuing the following commands: Action: Originate Context: default Exten: 5555555 Channel: local/1234 Priority: 1 Now, the phone rings at the agent side and it is being picked up. The other party answers and the call is set up. This is the output from the console: ============================================================================ -- Executing Queue("Local/[EMAIL PROTECTED],2", "managerqueue||||60") in new stack -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2 -- outgoing agentcall, to agent '1020', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1020 -- outgoing agentcall, to agent '1010', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1010 -- outgoing agentcall, to agent '1000', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1000 -- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/1020") in new stack Jun 4 18:47:22 NOTICE[12633]: app_dial.c:968 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) -- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/1010") in new stack -- Called 1010 -- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/1000") in new stack -- SIP/1010-a2f8 is ringing -- Agent/1010 is ringing -- Called 1000 -- SIP/1000-962b is ringing -- Agent/1000 is ringing -- SIP/1000-962b answered Local/[EMAIL PROTECTED],2 -- Agent/1000 answered Local/[EMAIL PROTECTED],2 -- Playing 'queue-reporthold' (language 'en') == Spawn extension (default, 1010, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (default, 1000, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- Playing 'queue-less-than' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'queue-minutes' (language 'en') -- Stopped music on hold on Local/[EMAIL PROTECTED],2 > Channel Local/[EMAIL PROTECTED],1 was answered. -- Executing SetVar("Local/[EMAIL PROTECTED],1", "__RECSUFFIX=5555555") in new stack -- Executing Dial("Local/[EMAIL PROTECTED],1", "SIP/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] ============================================================================ So far everything is normal. However (and this is the strange part), after completing this call, the agent answering the call becomes unavailable. Not logged out, just unavailable. And so he remains. Issuing the same actions from the asterisk manager interface again will result in the call going nowhere (if there's no other agents logged in), or to another agent, even though the queue has wrapuptime=0 and strategy=ringall. Even calling directly to the queue extension from an outside line will result in the same. After a little while, the system becomes unable to dial out, I get these messages: ============================================================================ Jun 4 18:52:12 WARNING[12633]: channel.c:341 ast_channel_alloc: Channel allocation failed: Refusing due to active shutdown Jun 4 18:52:12 WARNING[12633]: channel.c:341 ast_channel_alloc: Channel allocation failed: Refusing due to active shutdown Jun 4 18:52:12 WARNING[12633]: chan_local.c:498 local_new: Unable to allocate channel structure Jun 4 18:52:12 NOTICE[12633]: channel.c:1982 __ast_request_and_dial: Unable to request channel local/1234 Jun 4 18:52:12 WARNING[12633]: channel.c:341 ast_channel_alloc: Channel allocation failed: Refusing due to active shutdown Jun 4 18:52:12 WARNING[12633]: pbx.c:4778 ast_pbx_outgoing_cdr_failed: Unable to allocate channel structure for CDR record Jun 4 18:52:34 WARNING[12633]: channel.c:341 ast_channel_alloc: Channel allocation failed: Refusing due to active shutdown Jun 4 18:52:34 WARNING[12633]: chan_sip.c:2342 sip_new: Unable to allocate SIP channel structure Jun 4 18:52:34 NOTICE[12633]: chan_sip.c:8812 handle_request_invite: Unable to create/find channel ============================================================================ Also, after this, asterisk takes very long to stop with the stop now command, and when it finally exits, it returns memory segmentation error (or similar, translated to English), in addition to the well-known Yuck: Error in buffer handling...: Broken pipe Can anyone confirm this bug? I should add that originating a call not going into a queue, but directly to another extension works fine. Regards, Bjorn -----Opprinnelig melding----- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P� vegne av Tony Mountifield Sendt: 4. juni 2005 11:36 Til: [email protected] Emne: [Asterisk-Users] Re: Setting up calls through the manager interface In article <[EMAIL PROTECTED]>, Bjorn <[EMAIL PROTECTED]> wrote: > -=-=-=-=-=- > > Hello all! > > I am currently making a script which is supposed to set up a call on request from a user, > say, through a web page, for support issues etc. I am new into both asterisk and php, but I > am working my way through the path as good as I can. > > Basically, what I would want to do, is to give the user the possibility to initiate a call > by clicking a button. I?ve seen a cgi-alternative for this, but I would prefer it in PHP, > furthermore, extend the functionality of this a bit: > > Imagine, the user clicks a button to initiate a call. The script is called and establish a > connection to the manager interface. So far so good. > > The script will first call a support representative on the inside, and, when answered, it > will proceed with calling the customer. If there was only one support representative, this > could easily have been accomplished by executing the following: > > action: originate > context: local > exten: 555-4343 > priority: 1 > channel: SIP/1234 > > ? where channel would be the support rep.?s number. However, when there?s more than one, > you?d prefer to have the calls routed to whoever is available. This is nicely fixed in the > queue system, where the support representatives can log on and off, the calls goes to first > available representative etc. I suppose two alternatives would be the most common ones here, > to have the phone ring at all available channels within a ?support group? at the same time, > or have the call distributed randomly and (preferably) transferred to another agent if it > turns out there was no answer at the first representative. > > I tried to achieve this by the following: > > > action: originate > context: local > exten: 555-4343 > priority: 1 > channel: SIP/1234 > channel: SIP/2345 No, you need to use the "Local" channel type. So as not to confuse between labels I'll use a different name for the contexts. So in extensions.conf you might have something like this: [internal] exten => _1XXX,1,Dial(SIP/${EXTEN}) exten => _1XXX,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _1XXX,3,Hangup [external] exten => _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten => _X.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _X.,3,Hangup Then if you wanted to dial your internal extension of 1234 and when answered connect them to the outside number 0123456789, through the manager API you do the following: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: external Exten: 0123456789 Priority: 1 See http://www.voip-info.org/wiki-Asterisk+Local+channels for more info. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
