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Стратегии распределения звонков в очереди. [закрыт]

0

Подскажите пожалуйста по каким причинам могут не работать все возможные стратегии распределения звонков кроме ringall? Когда выбрана стратегия отличная от ringall звонок просто "висит" в постоянном ожидании.

Вот log звонка:

  trixbox1*CLI> #####################
<--- SIP read from UDP://192.168.33.188:53819 --->
ACK sip:699@192.168.33.79:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-771515539051727a-1---d8754z-;rport
Max-Forwards: 70
To: <sip:699@192.168.33.79:5060>;tag=as500ff939
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---########
trixbox1*CLI> #####################
<--- SIP read from UDP://192.168.33.188:53819 --->
INVITE sip:699@192.168.33.79:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-b871b55a147f843b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:600@192.168.33.188:53819;rinstance=52f1d5452aad7139>
To: <sip:699@192.168.33.79:5060>
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username="600",realm="asterisk",nonce="5b988018",uri="sip:699@192.168.33.79:5060",response="a8a96b362fac27d3c232814cae98f5f2",algorithm=MD5
Content-Length: 410

v=0
o=3cxVCE 69193725 351239940 IN IP4 192.168.33.188
s=3cxVCE Audio Call
c=IN IP4 192.168.33.188
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 192.168.33.188
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv

<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.33.188 : 53819 (NAT)
Using INVITE request as basis request - OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
Found user '600' for '600'
Found RTP audio format 0###########
Found RTP audio format 8###########
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x8000c (ulaw|alaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.33.188:40036
Peer video RTP is at port 192.168.33.188:40034
[Oct 30 19:44:00] DEBUG[2607]: pbx.c:3198 ast_hint_extension: FONALITY: This thread has already held the conlock, skip locking
Looking for 699 in from-internal (domain 192.168.33.79)
list_route: hop: <sip:600@192.168.33.188:53819;rinstance=52f1d5452aad7139>
trixbox1*CLI> #####################
<--- Transmitting (NAT) to 192.168.33.188:53819 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-b871b55a147f843b-1---d8754z-;received=192.168.33.188;rport=53819
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
To: <sip:699@192.168.33.79:5060>
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:699@192.168.33.79>
Content-Length: 0


<------------>
    -- Executing [699@from-internal:1] Macro("SIP/600-00000013", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/600-00000013", "AMPUSER=600") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/600-00000013", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/600-00000013", "1?Set(REALCALLERIDNUM=600)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/600-00000013", "AMPUSER=600") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/600-00000013", "AMPUSERCIDNAME=test600") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/600-00000013", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/600-00000013", "AMPUSERCID=600") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/600-00000013", "CALLERID(all)="test600" <600>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/600-00000013", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/600-00000013", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/600-00000013", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/600-00000013", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/600-00000013", "Using CallerID "test600" <600>") in new stack
    -- Executing [699@from-internal:2] Answer("SIP/600-00000013", "") in new stack
Audio is at 192.168.33.79 port 11824
Video is at 192.168.33.79 port 19164
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP#####
Adding video codec 0x80000 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.33.188:53819 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-b871b55a147f843b-1---d8754z-;received=192.168.33.188;rport=53819
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
To: <sip:699@192.168.33.79:5060>;tag=as77a46806
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:699@192.168.33.79>
Content-Type: application/sdp
Content-Length: 372

v=0
o=root 933844023 933844023 IN IP4 192.168.33.79
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 192.168.33.79
b=CT:384
t=0 0
m=audio 11824 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 19164 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

<------------>
    -- Executing [699@from-internal:3] Set("SIP/600-00000013", "__BLKVM_OVERRIDE=BLKVM/699/SIP/600-00000013") in new stack
    -- Executing [699@from-internal:4] Set("SIP/600-00000013", "__BLKVM_BASE=699") in new stack
    -- Executing [699@from-internal:5] Set("SIP/600-00000013", "DB(BLKVM/699/SIP/600-00000013)=TRUE") in new stack
    -- Executing [699@from-internal:6] ExecIf("SIP/600-00000013", "1?Set(_DIAL_OPTIONS=trM(auto-blkvm))") in new stack
    -- Executing [699@from-internal:7] Set("SIP/600-00000013", "__NODEST=699") in new stack
    -- Executing [699@from-internal:8] Set("SIP/600-00000013", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q699-20121030-194400-1351615440.23") in new stack
    -- Executing [699@from-internal:9] Set("SIP/600-00000013", "__MOHCLASS=VolvoWaiting") in new stack
    -- Executing [699@from-internal:10] Queue("SIP/600-00000013", "699,t,,") in new stack
    -- Started music on hold, class 'VolvoWaiting', on SIP/600-00000013
[Oct 30 19:44:00] NOTICE[3720]: app_queue.c:2785 wait_for_answer: No one is answering queue '699' (5/0/0)
trixbox1*CLI> #####################
<--- SIP read from UDP://192.168.33.188:53819 --->
ACK sip:699@192.168.33.79 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-25053d29e80e7377-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:600@192.168.33.188:53819;rinstance=52f1d5452aad7139>
To: <sip:699@192.168.33.79:5060>;tag=as77a46806
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 2 ACK
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username="600",realm="asterisk",nonce="5b988018",uri="sip:699@192.168.33.79:5060",response="a8a96b362fac27d3c232814cae98f5f2",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
trixbox1*CLI> #####################
<--- SIP read from UDP://192.168.33.188:53819 --->
BYE sip:699@192.168.33.79 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-5e692319ec748761-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:600@192.168.33.188:53819;rinstance=52f1d5452aad7139>
To: <sip:699@192.168.33.79:5060>;tag=as77a46806
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 3 BYE
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username="600",realm="asterisk",nonce="5b988018",uri="sip:699@192.168.33.79",response="b246c9cc344f685a2237406bdf7baf1e",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.33.188 : 53819 (NAT)

<--- Transmitting (NAT) to 192.168.33.188:53819 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.188:53819;branch=z9hG4bK-d8754z-5e692319ec748761-1---d8754z-;received=192.168.33.188;rport=53819
From: "test600"<sip:600@192.168.33.79:5060>;tag=6731d17e
To: <sip:699@192.168.33.79:5060>;tag=as77a46806
Call-ID: OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.
CSeq: 3 BYE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Stopped music on hold on SIP/600-00000013
  == Spawn extension (from-internal, 699, 10) exited non-zero on 'SIP/600-00000013'
    -- Executing [h@from-internal:1] Macro("SIP/600-00000013", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/600-00000013", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/600-00000013", "0?skipblkvm") in new stack
    -- Executing [s@macro-hangupcall:5] NoOp("SIP/600-00000013", "Cleaning Up Block VM Flag: BLKVM/699/SIP/600-00000013") in new stack
    -- Executing [s@macro-hangupcall:6] DBdel("SIP/600-00000013", "BLKVM/699/SIP/600-00000013") in new stack
    -- DBdel: family=BLKVM, key=699/SIP/600-00000013
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/600-00000013", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)#
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/600-00000013", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/600-00000013' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/600-00000013'
[Oct 30 19:44:05] DEBUG[2607]: pbx.c:3198 ast_hint_extension: FONALITY: This thread has already held the conlock, skip locking
Really destroying SIP dialog 'OWM4YzBlOTA2ODhlZDcyMzc0NzQ4OWRjNjc0MGI0ZjI.' Method: BYE
trixbox1*CLI> #####################

Вот файл queue_additional.conf, т.к. настраивал очередь через Trixbox:

[699]
announce-frequency=0
announce-holdtime=no
announce-position=no
autofill=yes
eventmemberstatus=yes
eventwhencalled=yes
joinempty=yes
leavewhenempty=no
maxlen=0
monitor-type=mixmonitor
monitor-format=gsm
music=VolvoWaiting
periodic-announce-frequency=30
queue-callswaiting=silence/1
queue-thereare=silence/1
queue-youarenext=silence/1
retry=0
strategy=leastrecent
timeout=15
weight=1
wrapuptime=0
context=ivr-6
periodic-announce=custom/busy2_volvo_nomusic
member=Local/601@from-internal/n,0
member=Local/602@from-internal/n,0
member=Local/603@from-internal/n,0
member=Local/604@from-internal/n,0
member=Local/600@from-internal/n,0

queue show:

trixbox1*CLI> queue show
default has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   No Members
   No Callers

agents has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   No Members
   No Callers

699 has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime), W:1, C:0, A:1, SL:0.0% within 0s
   Members:
      Local/605@from-internal/n (Not in use) has taken no calls yet with total talktime 0s
      Local/603@from-internal/n (Not in use) has taken no calls yet with total talktime 0s
      Local/602@from-internal/n (Not in use) has taken no calls yet with total talktime 0s
      Local/601@from-internal/n (Not in use) has taken no calls yet with total talktime 0s
      Local/600@from-internal/n (Not in use) has taken no calls yet with total talktime 0s
   No Callers
удалить переоткрыть спам изменить тег редактировать

спросил 2012-10-26 14:37:51 +0400

dont_panic Gravatar dont_panic
1 1 3
http://www.dmbasis.ru/

обновил 2012-10-30 20:55:36 +0400

Comments

покажите логи

awsswa ( 2012-10-26 15:07:18 +0400 )редактировать

2 Ответа

0

Скорее всего нет ни одного зарегистрированного агента. agent show online и queue show что показывают ?

ссылка удалить спам редактировать

ответил 2012-10-26 18:01:45 +0400

amonra Gravatar amonra flag of Ukraine
2261 24 13 63
http://lantec.ua/
0

по причине неправильно выставленной priority например.

но без конфигов и логов это гадание.

ссылка удалить спам редактировать

ответил 2012-10-27 18:14:49 +0400

meral Gravatar meral flag of Ukraine
21707 24 20 172
http://pro-sip.net/

Comments

А можно подробнее по поводу priority? Как правильно он должен быть выставлен? А то больше как-то не на что думать.

dont_panic ( 2012-10-30 16:15:51 +0400 )редактировать

к вашему случаю похоже не относится. вот идите сюда http://asterisk-support.ru/question/38386/kak-poniat-chto-proiskhodit-na-asteriske/ включайте дебаг и сделайте queue show

meral ( 2012-10-30 19:37:27 +0400 )редактировать

Добавил. лог звонка тоже нужен?

dont_panic ( 2012-10-30 20:28:20 +0400 )редактировать

надо дебаг. с лога ниего не понятн. или он не полный.

meral ( 2012-10-30 20:34:37 +0400 )редактировать

call-limit=2

awsswa ( 2012-10-30 20:36:01 +0400 )редактировать

Добавил debug. awsswa, простите мне мою недалёкость, но что вы подразумевали под "call-limit=2"

dont_panic ( 2012-10-30 20:58:25 +0400 )редактировать

выставьте на каждого кто в очереди состоит call-limit=2, начинайте осваивать tcpdump и выкладывать дебаг из него

awsswa ( 2012-10-30 21:06:01 +0400 )редактировать

хз.зачем вы сип дебаг включили? общий включайте. или наймите когото.

meral ( 2012-10-30 21:06:35 +0400 )редактировать

core set debug 9 и core set verbose 99

awsswa ( 2012-10-30 21:09:09 +0400 )редактировать

файл получается очень длинный. Скажите, с этой проблемой кто-нибудь раньше сталкивался?

dont_panic ( 2012-10-30 21:24:34 +0400 )редактировать
1

Сделайте тестовую очередь на двух - что по меньше был

awsswa ( 2012-10-30 21:53:12 +0400 )редактировать
2

у меня машина не едет. заводить сильно долго. скажиет что починить? кто сталкивалься?

meral ( 2012-10-30 22:15:33 +0400 )редактировать

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Статистика

Задан: 2012-10-26 14:37:51 +0400

Просмотрен: 430 раз

Обновлен: Oct 30 '12

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.