VoiceMail и запись сообщения

[strange]

Дефолтное, честное слово. Ничего не менял. Права тоже сейчас проверил, все есть. Владелец тоже везде asterisk.

[awsswa]

показывайте voicemail.conf
Если не найдем ничего криминального - только обновляться до текущей версии

[Vlad1983]

"Что то ответило 200, после ошибки."
это мегафон на BYE от астериск
смотрите выше

мегафон -- asterisk
INVITE -->
<-- RING
<-- 183 ; возможно не будет
<-- 200 OK

[strange]

root@ats:/var/spool/asterisk/voicemail/default# cat /etc/asterisk/voicemail.conf
[general]
format=wav
serveremail=ats@kdau.ru
attach=yes
maxmsg=100
maxsecs=180
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
emaildateformat=%A, %B %d, %Y at %r
pagerdateformat=%A, %B %d, %Y at %r
mailcmd=/usr/sbin/ssmtp -t
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only]
; This is intended for use with users who wish to receive their
; voicemail ONLY by email. Note: "deletevoicemail" is provided as an
; equivalent option for Realtime configuration.
[zonemessages]


[default]
; почтовыйящик => пароль, имя[,email[,email_пейджера[,опции]]]
5001 => 9999,ivan,info@xxxxxx.ru
5002 => 9999,strange,sysadminrus@xxxxx.com

[other]
;
;[acme]
;
;[imapvm]
;4324 => 7764,Ellis Redding,red@buxton.us,,imapuser=eredding|imappassword=g3tbusy|imapfolder=notinbox
;4325 => 2392,Andrew Dufresne,andy@dufresne.info,,imapuser=adufresne|imappassword=rockh@mmer

[strange]

Выложил всё, к сожалению не смог понять, что конкретно требуется.

REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/UDP 188.168.202.148:5060;branch=z9hG4bK50aa4c68;rport
Max-Forwards: 70
From: <sip:79221882778@multifon.ru>;tag=as05ac2faf
To: <sip:79221882778@multifon.ru>
Call-ID: 6319cfa81104267d457931c8762f73fb@127.0.1.1
CSeq: 121 REGISTER
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="79221882778", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTM1OTQ1MDg0MzpAxDaL8kVmbt52XC58Jinp", response="83e425809088aaf88e2d5c6ff4e8f9d7", opaque="MTM1OTQ1MDg0MzpAxDaL8kVmbt52XC58Jinp", qop=auth, cnonce="653ba1dd", nc=00000013
Expires: 120
Contact: <sip:79221882778@188.168.202.148:5060>
Content-Length: 0


---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.229:5060;received=188.168.202.148;branch=z9hG4bK50aa4c68;rport=5060
From: <sip:79221882778@multifon.ru>;tag=as05ac2faf
To: <sip:79221882778@multifon.ru>;tag=aprqp67ca23-6d6o8c10000i7
Call-ID: 6319cfa81104267d457931c8762f73fb@127.0.1.1
CSeq: 121 REGISTER
P-Associated-URI:
Contact: <sip:79221882778@192.168.0.229:5060>;expires=168
Service-Route: <sip:79221882778@193.201.229.35:5060;transport=udp;lr>

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '6319cfa81104267d457931c8762f73fb@127.0.1.1' in 32000 ms (Method: REGISTER)
[Jan 29 15:45:45] NOTICE[1000]: chan_sip.c:20714 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '6319cfa81104267d457931c8762f73fb@127.0.1.1' Method: REGISTER

<--- SIP read from UDP:193.201.229.35:5060 --->
INVITE sip:79221882778@192.168.0.229:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK2t628l3048ggd7b4b681.1
Max-Forwards: 19
From: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d26b27ecd
User-Agent: Dialogic-SIP/10.5.3.203 IMG1 0
Allow: PRACK,INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Supported: 100rel,path,replaces,tdialog
Expires: 180
Organization: Dialogic
Content-Length: 293
Content-Type: application/sdp
To: sip:79221882778-dcfceti8137e4@10.190.35.4:5060
Call-ID: 0203205E31814000000185DF@SFESIP2-id1-ext
CSeq: 1 INVITE
Contact: <sip:73433734397@193.201.229.35:5060;transport=udp>

v=0
o=Dialogic_SDP 1187589 0 IN IP4 193.201.229.35
s=Dialogic-SIP
c=IN IP4 193.201.229.35
t=0 0
m=audio 23430 RTP/AVP 8 0 18 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=silenceSupp:off - - - -
<------------->
--- (15 headers 13 lines) ---
Sending to 193.201.229.35:5060 (NAT)
Using INVITE request as basis request - 0203205E31814000000185DF@SFESIP2-id1-ext
Found peer 'multifon' for '73433734397' from 193.201.229.35:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 193.201.229.35:23430
Looking for 79221882778 in incoming (domain 192.168.0.229)
list_route: hop: <sip:73433734397@193.201.229.35:5060;transport=udp>

<--- Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK2t628l3048ggd7b4b681.1;received=193.201.229.35;rport=5060
From: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d26b27ecd
To: sip:79221882778-dcfceti8137e4@10.190.35.4:5060
Call-ID: 0203205E31814000000185DF@SFESIP2-id1-ext
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:79221882778@188.168.202.148:5060>
Content-Length: 0


<------------>
-- Executing [79221882778@incoming:1] Wait("SIP/multifon-00000008", "15") in new stack
-- Executing [79221882778@incoming:2] Playback("SIP/multifon-00000008", "/home/music/output") in new stack
Audio is at 11412
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK2t628l3048ggd7b4b681.1;received=193.201.229.35;rport=5060
From: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d26b27ecd
To: sip:79221882778-dcfceti8137e4@10.190.35.4:5060;tag=as187e4708
Call-ID: 0203205E31814000000185DF@SFESIP2-id1-ext
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:79221882778@188.168.202.148:5060>
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 1281491040 1281491040 IN IP4 188.168.202.148
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 188.168.202.148
t=0 0
m=audio 11412 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:193.201.229.35:5060 --->
ACK sip:79221882778@192.168.0.229:5060 SIP/2.0
Via: SIP/2.0/UDP 193.201.229.35:5060;branch=z9hG4bK8lfio3105oh09hvp55t1.1
Max-Forwards: 19
User-Agent: Dialogic-SIP/10.5.3.203 IMG1 0
Content-Length: 0
CSeq: 1 ACK
To: sip:79221882778-dcfceti8137e4@10.190.35.4:5060;tag=as187e4708
From: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d26b27ecd
Call-ID: 0203205E31814000000185DF@SFESIP2-id1-ext

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6319cfa81104267d457931c8762f73fb@127.0.1.1' Method: REGISTER
-- <SIP/multifon-00000008> Playing '/home/music/output.ulaw' (language 'en')
-- Executing [79221882778@incoming:3] Dial("SIP/multifon-00000008", "SIP/5001,10") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5001
-- SIP/5001-00000009 is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.0.41:5061
-- SIP/5001-00000009 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [79221882778@incoming:4] VoiceMail("SIP/multifon-00000008", "5001@default,u") in new stack
-- <SIP/multifon-00000008> Playing 'vm-theperson.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'digits/5.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'digits/0.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'digits/0.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'digits/1.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'vm-isunavail.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'vm-intro.slin' (language 'en')
-- <SIP/multifon-00000008> Playing 'beep.slin' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/kaeE3l format: wav, 0x7fc7d8019e78
[Jan 29 15:46:40] WARNING[6908]: app.c:855 __ast_play_and_record: No audio available on SIP/multifon-00000008??
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/5001/INBOX/msg0017.txt': == Found
== Spawn extension (incoming, 79221882778, 4) exited non-zero on 'SIP/multifon-00000008'
Scheduling destruction of SIP dialog '0203205E31814000000185DF@SFESIP2-id1-ext' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:73433734397@193.201.229.35:5060;transport=udp> for address/port to send to
set_destination: set destination to 193.201.229.35:5060
Reliably Transmitting (NAT) to 193.201.229.35:5060:
BYE sip:73433734397@193.201.229.35:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 188.168.202.148:5060;branch=z9hG4bK74388347;rport
Max-Forwards: 70
From: sip:79221882778-dcfceti8137e4@10.190.35.4:5060;tag=as187e4708
To: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d26b27ecd
Call-ID: 0203205E31814000000185DF@SFESIP2-id1-ext
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.229:5060;received=188.168.202.148;branch=z9hG4bK74388347;rport=5060
From: sip:79221882778-dcfceti8137e4@10.190.35.4:5060;tag=as187e4708
To: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d26b27ecd
Call-ID: 0203205E31814000000185DF@SFESIP2-id1-ext
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '0203205E31814000000185DF@SFESIP2-id1-ext' Method: ACK

[Vlad1983]

в точки зрения сигналки нормально
собирайте руками

[strange]

Благодарю всех за ответы.
Последний вопрос, какую версию вы посоветовали бы скачать?
Вижу 11LTS
1.8 LTS
Есть ещё какой-то Certified 1.8 LTS

[awsswa]

Последная из исходников 1.8.20.1
Только на ubuntu запросто не обновиться, там dahdi сносить надо полностью и ставить по новой 2.6.1

[strange]

Собрался без проблем. Но ошибка никуда не делась.
root@ats:/home/strange# asterisk -V
Asterisk 1.8.20.1
root@ats:/home/strange# asterisk -rvvv
Asterisk 1.8.20.1, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.20.1 currently running on ats (pid = 990)
Verbosity was 0 and is now 3
== Using SIP RTP CoS mark 5
-- Executing [79221882778@incoming:1] Wait("SIP/multifon-00000000", "15") in new stack
-- Executing [79221882778@incoming:2] Playback("SIP/multifon-00000000", "/home/music/output") in new stack
-- <SIP/multifon-00000000> Playing '/home/music/output.ulaw' (language 'en')
-- Executing [79221882778@incoming:3] Dial("SIP/multifon-00000000", "SIP/5001,10") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5001
-- SIP/5001-00000001 is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.0.41:5061
-- SIP/5001-00000001 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [79221882778@incoming:4] VoiceMail("SIP/multifon-00000000", "5001@default,u") in new stack
-- <SIP/multifon-00000000> Playing 'vm-theperson.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/5.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/0.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/0.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/1.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'vm-isunavail.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'vm-intro.gsm' (language 'en')
-- <SIP/multifon-00000000> Playing 'beep.gsm' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/V1k7TO format: wav, 0x7f36a8018048
[Jan 29 16:48:17] WARNING[1471]: app.c:860 __ast_play_and_record: No audio available on SIP/multifon-00000000??
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/5001/INBOX/msg0020.txt': == Found
== Spawn extension (incoming, 79221882778, 4) exited non-zero on 'SIP/multifon-00000000'


Может быть у кого-то какие-то предположения остались?

[Vlad1983]

последний шанс
ps axu | grep asterisk
допустим вывалит так:
...
asterisk 2777 0.4 0.2 712416 18952 ? Ssl 2012 337:01 /usr/sbin/asterisk
...
тогда правим
chown asterisk:asterisk -R /var/spool/asterisk/

если нифига стучите в скайп бум разбираться (бесплатно, т.к. самому аж интересно, что там можно было наворотить такого)