asterisk.ru

Добрый день Уважаемые Форумчане!
Прошу помощи в решении такой проблемы:
Есть определенная очередь входящих звонков:

[general-queue]
announce-frequency=0
announce-holdtime=no
eventmemberstatus=no
eventwhencalled=no
joinempty=yes
leavewhenempty=no
maxlen=0
music=default
periodic-announce-frequency=0
queue-callswaiting=silence/1
queue-thereare=silence/1
queue-youarenext=silence/1
retry=5
strategy=ringall
timeout=600
wrapuptime=0
member=SIP/701
member=SIP/710
member=SIP/715
member=SIP/717
member=SIP/719
member=SIP/720

При входящем звонке когда звонок попадает в очередь и если хоть один из номеров не зарегистрирован в трубку говорит - "Номер не подключен" и ложится трубка.
Как исправить данную ситуацию, что бы при входящем просто игнорировался незарегистрированный добавочный и звонки шли на оставшихся членов очереди?

joinempty=no
leavewhenempty=yes

Сделал как Вы посоветовали.
К сожалению проблему это не решило. В лог вываливается такое сообщение:

[2013-05-31 13:09:35] WARNING[19289]: chan_sip.c:17016 handle_response: Remote host can't match request CANCEL to call '245c48777124ae50445ce4dc2df73558@192.168.4.254'. Giving up.

покажите
asterisk -rx "queue show"

asterisk -rx "queue show"
general-queu has 0 calls (max unlimited) in 'ringall' strategy (2s holdtime), W: 0, C:3, A:0, SL:0.0% within 0s
Members:
SIP/701 (Not in use) has taken no calls yet
SIP/715 (Not in use) has taken 2 calls (last was 83 secs ago)
SIP/717 (Not in use) has taken 1 calls (last was 768 secs ago)
SIP/719 (Not in use) has taken no calls yet
SIP/720 (Not in use) has taken no calls yet
No Callers

710 я закоментировал, что бы заработала телефония и поэтому он тут не отображается...

добавте
autofill=yes

если добавить не зареганого что в
asterisk -rx "queue show"
?

timeout = 600 - много
от 8 до 20 обычно

Добавил
autofill=yes - ситуацию решить не помогло

При раскоментированой строке вывод очереди такой:

asterisk -rx "queue show"
general-queu has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/701 (Not in use) has taken no calls yet
SIP/710 (Not in use) has taken no calls yet
SIP/715 (Not in use) has taken no calls yet
SIP/717 (Not in use) has taken no calls yet
SIP/719 (Not in use) has taken no calls yet
SIP/720 (Not in use) has taken no calls yet
No Callers

asterisk -rx "sip show settings"
asterisk -rx "sip show peer 710"

Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: 5060
TCP Bindaddress: 0.0.0.0
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
AutoCreate Peer: No
Match Auth Username: Yes
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Call limit peers only: No
Direct RTP setup: No
User Agent: Asterisk PBX
SDP Session Name: Asterisk PBX 1.6.0.9
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: On
T38 fax pt UDPTL: No
SIP realtime: Disabled
Qualify Freq : 60000 ms

Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 127.0.1.1:5060
STUN server: 0.0.0.0:0

Global Signalling Settings:
---------------------------
Codecs: 0x3c010d (g723|ulaw|alaw|g729|h261|h263|h263p|h264)
Codec Order: alaw:20,ulaw:20,g723:30,g729:20
Relax DTMF: Yes
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 1800 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000

Default Settings:
-----------------
Context: incoming
Nat: Always
DTMF: auto
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: ru
MOH Interpret: default
MOH Suggest: default
Voice Mail Extension: asterisk

----




* Name : 710
Secret : <Set>
MD5Secret : <Not set>
Context : long
Subscr.Cont. : <Not set>
Language : ru
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "phone710" <710>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Nat : Always
ACL : Yes
T38 pt UDPTL : Yes
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : Yes
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : auto
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : (Unspecified) Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Transport : UDP
Def. Username: 710
SIP Options : (none)
Codecs : 0x30d (g723|ulaw|alaw|g729|speex)
Codec Order : (alaw:20,ulaw:20,g723:30,g729:20,speex:20)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs