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HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070603/6564c117/attachment.htm

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcem...

Martin Pycko <martinp@digium.com> writes: > sure, use the 'n' option of the queue and put voicemail app as the next > priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to the head of the queue (inside the wait_our_turn function). Unless the ast_waitfordigit function checks the timeout and I missed it, this solution won't work. Am I reading the code right? > On Fr...

...after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing (WARNING[1863]: app_queue.c:3136 try_calling: The device state of this queue member, Agent/136, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings.) However, when i look at the queue a few sec after the Agent is marked 'in use' which wasn't the case with IA...

...xt action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse, if there are no additional actions for the extension, the call is disconnected. You gotta love the error message asterisk spits out when that happens "app_queue.c:1170 try_calling: Agent on Agent/7777 hungup on the customer. They're going to be pissed.". So here is my question, is it possible to avoid this? If not, is there a command to direct a call back into the queue, preferably at the front of the line? Thanks, ~Senyo -------------- next...

...; qe.last_pos_said = 0; qe.last_pos = 0; + qe.last_ring_time = 0; qe.last_periodic_announce_time = time(NULL); qe.last_periodic_announce_sound = 0; qe.valid_digits = 0; @@ -4074,9 +4080,12 @@ break; } /* Try calling all queue members for 'timeout' seconds */ - res = try_calling(&qe, args.options, args.announceoverride, args.url, &tries, &noption, args.agi); - if (res) - goto stop; + if ((time(NULL) - qe.last_ring_time) > qe.parent->retry) { + res = try_calling(&qe, args.options, args.announceoverride, args.url, &tries, &noption,...

...s=0xb51e2e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xa2db720, c1=0xa12acf8, config=0xb51e37c0, fo=0xb51e2f50, rc=0xb51e2f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xa2db720, peer=0xa12acf8, config=0xb51e37c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb51e3ac0, options=Variable "options" is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0xa2db720, data=0xb51e8010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0xa2db720, con=Variable "con" is not available. ) at pbx.c:574 #8...

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies DB I have a column "call-limit" which has...

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

...it=54 '6') at channel.c:2692 #3 0x0805d20d in ast_dtmf_stream (chan=0x925cb78, peer=0xb6a2d0e0, digits=0xb6940bf4 "6", between=100) at app.c:243 #4 0xb79086ca in ast_bridge_call (chan=0xb6a2d0e0, peer=0x925cb78, config=0xb69424d4) at res_features.c:1473 #5 0xb6e011ae in try_calling (qe=0xb6942784, options=0x925cb78 "@`\201?\2205&\b?N#\t\003O#\t(r)?\023\b(r)?\023\b(r)?\023\b?N#\t?N#\t\200", announceoverride=0xb694270e "", url=0xb694270d "", go_on=0xb6942770, agi=0x0) at app_queue.c:2651 #6 0xb6df9fda in queue_exec (chan=0xb6a2d0e0...

...s from *external Database thru ODBC*, using* Local channels * as interface - sip extensions from *external Database thru ODBC* When a call is sent from queue to an interface (local channel), it is answered but a message appears at the CLI: *[Nov 4 16:56:04] WARNING[13951]: app_queue.c:3014 try_calling: The device state of this queue member, Ag-Daniel, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. * The interface value for agent *Ag-Daniel* is: *Local/7070 at agent-channels/n*and final interface is *SIP/7070* I have...

...s] maxlogintries=3 musiconhold => default updatecdr=yes agent => 21,1234,Klaudia agent => 22,1234,Daniele agent => 23,1234,Daniela What about the wrapuptime in agents.conf - do i have to set this to the same as in queues.conf? i also get this message: WARNING[23115]: app_queue.c:3014 try_calling: The device state of this queue member, Agent/22, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. the Agent/22 is "not in use", there are no open channels and "queue show" is also reporting "not...

...is this the asterisk inner mechanism or is my configuration error? Thanks! messages on the cli as follow: -- SIP/1003-0000001d is ringing -- SIP/1003-0000001d answered SIP/1004-0000001c -- Stopped music on hold on SIP/1004-0000001c [Jan 18 10:08:42] WARNING[17022]: app_queue.c:3268 try_calling: The device state of this queue member, Zhang Jianming, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. -- Packet2Packet bridging SIP/1004-0000001c and SIP/1003-0000001d -- Executing Playback("SIP/1004-00000...

...0x009e55f1 in odbc_log (cdr=0xa4a7c88) at cdr_odbc.c:137 #12 0x0808db8a in post_cdr (cdr=0xa4a7c88) at cdr.c:1055 #13 0x0808e36e in ast_cdr_detach (cdr=0xa4a7c88) at cdr.c:1251 #14 0x080c042b in ast_bridge_call (chan=0xa52e8d0, peer=0xb79047a8, config=0x2abe1c0) at features.c:2849 #15 0x026285d9 in try_calling (qe=0x2abe6f0, options=Variable "options" is not available. ) at app_queue.c:4269 #16 0x0262d4a0 in queue_exec (chan=0xa52e8d0, data=0x2ac0a20) at app_queue.c:5204 #17 0x080f3c47 in pbx_exec (c=0xa52e8d0, app=0xb7c6b088, data=0x2ac0a20) at /usr/local/src/asterisk162/asterisk-1.6.2...

...t;Agent", format=2, data=0xb55fc9ba, cause=0xb55fcacc) at channel.c:3203 #17 0x00b62d13 in ring_entry (qe=0xb55fed00, tmp=0x98397f8, busies=0xb55fec44) at app_queue.c:1921 #18 0x00b639c0 in ring_one (qe=0xb55fed00, outgoing=0x9496d60, busies=0xb55fec44) at app_queue.c:2071 #19 0x00b6c670 in try_calling (qe=0xb55fed00, options=<value optimized out>, announceoverride=0x0, url=0x0, tries=0xb55feea0, noption=0xb55fee9c, agi=0x0) at app_queue.c:2960 #20 0x00b6fdca in queue_exec (chan=0x96e44b8, data=0xb5600f28) at app_queue.c:4083 #21 0x080ca6eb in pbx_extension_helper (c=0x96e44b8, con=0...

...4a4c68] asterisk channel.c:6447 ast_request() #13: [0x7f625d021eda] app_queue.so app_queue.c:4717 ring_entry() #14: [0x7f625d0232d3] app_queue.so app_queue.c:4886 ring_one() #15: [0x7f625d025592] app_queue.so app_queue.c:5487 wait_for_answer() #16: [0x7f625d026fa3] app_queue.so app_queue.c:7110 try_calling() #17: [0x7f625d02a4e1] app_queue.so app_queue.c:8730 queue_exec() #18: [0x53b599] asterisk pbx_app.c:493 pbx_exec() #19: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper() #20: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run() #21: [0x53184b] asterisk pbx.c:4669 decrease_call_count() #22...

...ported by Alexander Traud) * ASTERISK-25573 - [patch] H.264 format attribute module: resets whole SDP (Reported by Alexander Traud) * ASTERISK-24958 - Forwarding loop detection inhibits certain desirable scenarios (Reported by Mark Michelson) * ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked! (Reported by Alec Davis) * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by Joshua Colp) * ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally (Reported...

...ported by Alexander Traud) * ASTERISK-25573 - [patch] H.264 format attribute module: resets whole SDP (Reported by Alexander Traud) * ASTERISK-24958 - Forwarding loop detection inhibits certain desirable scenarios (Reported by Mark Michelson) * ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked! (Reported by Alec Davis) * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by Joshua Colp) * ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally (Reported...

...ported by Alexander Traud) * ASTERISK-25573 - [patch] H.264 format attribute module: resets whole SDP (Reported by Alexander Traud) * ASTERISK-24958 - Forwarding loop detection inhibits certain desirable scenarios (Reported by Mark Michelson) * ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked! (Reported by Alec Davis) * ASTERISK-25565 - DNS: System resolver only returns 1 record per result (Reported by George Joseph) * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by Joshua Co...