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