 | 1 | изначальная версия редактировать | |
Debian 7.1, Asterisk 1.6.0. SIP-транк от билайн, статический ип, eth1. SIP-транк comtube идет через eth0, как и интернет. Используется только для звонков в европу, номера, начинающиеся с 810.
Периодически на билайне пропадает слышимость собеседника,т.е. входящий звук. Непонятно в связи с чем. На Comtube все ок, никаких проблем.
Вот лог во время звонка:
<------------>
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Audio is at 10.79.87.94 port 14390
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x8 (alaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x4 (ulaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (NAT) to 10.225.61.130:5060:
INVITE sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
User-Agent: Topsail Asterisk PBX
Date: Tue, 16 Dec 2014 20:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 351041744 351041744 IN IP4 10.79.87.94
s=Asterisk PBX SVN-branch-1.6.0-r347660
c=IN IP4 10.79.87.94
t=0 0
m=audio 14390 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (6 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Got RTP packet from 11.12.13.5:10002 (type 08, seq 002308, ts 223115431, len 000160)
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
Reason: Q.850;cause=31;text="Normal, unspecified"
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Transmitting (NAT) to 10.225.61.130:5060:
ACK sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 ACK
User-Agent: Topsail Asterisk PBX
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Scheduling destruction of SIP dialog '139841780521131-221003236725599@11.12.13.5' in 32000 ms (Method: ACK)
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: Parsing <sip:11@11.12.13.5:5060> for address/port to send to
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: set destination to 11.12.13.5, port 5060
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (no NAT) to 11.12.13.5:5060:
BYE sip:11@11.12.13.5:5060 SIP/2.0
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
Max-Forwards: 70
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
User-Agent: Topsail Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
Server: Voip Phone 1.0
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: SIP Response message for INCOMING dialog BYE arrived
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '0ec920f346173cda626bbcc961afde88@10.225.61.130' Method: INVITE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '289305030e5e402416850bc957a1ae23@10.225.61.130' Method: BYE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '139841780521131-221003236725599@11.12.13.5' Method: ACK
[Dec 16 23:46:41] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.7:5060 --->
alive
<------------->
Постоянно идет инвайт на непонятный номер h.
sip.conf:
register => ид:пароль@sip.comtube.com/ид
;register => хххххх@10.225.61.130/хххххх
[beeline]
insecure=Invite,port
type=friend
fromuser=хххххх
canreinvite=no
context=from_beeline
qualify=yes
host=10.225.61.130
trunkname=beeline
dtmfmode=auto
nat=yes
srvlookup=yes
fromdomain=10.225.61.130
hassip=yes
dtmfmode=auto
[comtube]
username=хххххх
type=friend
secret=пароль
nat=yes
insecure=Invite,port
context=from_comtube
host=sip.comtube.com
trunkname=comtube
hassip=yes
fromuser=хххххх
fromdomain=sip.comtube.com
dtmfmode=auto
canreinvite=no
qualify=yes
extensions.conf:
[office]
exten => 11, 1, Dial(SIP/11,30)
;exten => 11, n, Playback(vmnobodyavail)
;exten => 11, n, Hangup()
exten => 12, 1, Dial(SIP/12,30)
;exten => 12, n, Playback(vmnobodyavail)
;exten => 12, n, Hangup()
exten => 13, 1, Dial(SIP/13,30)
;exten => 13, n, Playback(vmnobodyavail)
;exten => 13, n, Hangup()
exten => 14, 1, Dial(SIP/14,30)
;exten => 14, n, Playback(vmnobodyavail)
;exten => 14, n, Hangup()
;include => comtube_outbound
include => beeline_outbound
[comtube_outbound]
exten => _810.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _890.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _.{11},2,PlayBack(noanswer)
; exten => _.{11},4,HangUp
; exten => _.{11},5,PlayBack(busy)
; exten => _.{11},6,HangUp
[from_comtube]
exten => 775236,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 775236,2,Dial(SIP/11&SIP/12&SIP/13,25,Ttr) ; Входящие перенаправляются на внут$
exten => 775236,3,Hangup
[beeline_outbound]
exten => _.{6},1,Dial(SIP/beeline/${EXTEN},120)
;exten => _.{6},2,PlayBack(noanswer)
;exten => _.{6},4,HangUp
;exten => _.{6},5,PlayBack(busy)
;exten => _.{6},6,HangUp
[from_beeline]
exten => 313029,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 313029,2,Dial(SIP/11&SIP/12&SIP/13&SIP/14,25,Ttr) ; Входящие перенаправляются на внут$
exten => 313029,3,Hangup
Помогите понять, где я накосячил.
| | 2 | уточнение редактировать |
Debian 7.1, Asterisk 1.6.0.
SIP-транк от билайн, билайна включен в отдельную сетевуху eth1, статический ип, eth1.
ип.
SIP-транк comtube идет через eth0, как и интернет. Используется роутер, раздающий на офисную локалку интернет от билайна, тоже статический ип. Comtube используется только для звонков в европу, на номера, начинающиеся с 810.
Периодически на билайне пропадает слышимость собеседника,т.е. входящий звук. собеседника, т.е. не слышно собеседника. Непонятно в связи с чем. На Comtube все ок, никаких проблем.проблем.
externip=10.79.87.94 стоит ип сетевухи eth1.
Вот лог во время звонка:звонка на номер 060 (внешний):
<------------>
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Audio is at 10.79.87.94 port 14390
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x8 (alaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x4 (ulaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (NAT) to 10.225.61.130:5060:
INVITE sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
User-Agent: Topsail Asterisk PBX
Date: Tue, 16 Dec 2014 20:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 351041744 351041744 IN IP4 10.79.87.94
s=Asterisk PBX SVN-branch-1.6.0-r347660
c=IN IP4 10.79.87.94
t=0 0
m=audio 14390 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (6 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Got RTP packet from 11.12.13.5:10002 (type 08, seq 002308, ts 223115431, len 000160)
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
Reason: Q.850;cause=31;text="Normal, unspecified"
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Transmitting (NAT) to 10.225.61.130:5060:
ACK sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 ACK
User-Agent: Topsail Asterisk PBX
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Scheduling destruction of SIP dialog '139841780521131-221003236725599@11.12.13.5' in 32000 ms (Method: ACK)
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: Parsing <sip:11@11.12.13.5:5060> for address/port to send to
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: set destination to 11.12.13.5, port 5060
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (no NAT) to 11.12.13.5:5060:
BYE sip:11@11.12.13.5:5060 SIP/2.0
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
Max-Forwards: 70
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
User-Agent: Topsail Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
Server: Voip Phone 1.0
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: SIP Response message for INCOMING dialog BYE arrived
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '0ec920f346173cda626bbcc961afde88@10.225.61.130' Method: INVITE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '289305030e5e402416850bc957a1ae23@10.225.61.130' Method: BYE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '139841780521131-221003236725599@11.12.13.5' Method: ACK
[Dec 16 23:46:41] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.7:5060 --->
alive
<------------->
Постоянно идет инвайт на непонятный номер h.
sip.conf:
register => ид:пароль@sip.comtube.com/ид
;register => хххххх@10.225.61.130/хххххх
313029@10.225.61.130/313029
[beeline]
insecure=Invite,port
type=friend
fromuser=хххххх
fromuser=313029
canreinvite=no
context=from_beeline
qualify=yes
host=10.225.61.130
trunkname=beeline
dtmfmode=auto
nat=yes
srvlookup=yes
fromdomain=10.225.61.130
hassip=yes
dtmfmode=auto
[comtube]
username=хххххх
type=friend
secret=пароль
nat=yes
insecure=Invite,port
context=from_comtube
host=sip.comtube.com
trunkname=comtube
hassip=yes
fromuser=хххххх
fromdomain=sip.comtube.com
dtmfmode=auto
canreinvite=no
qualify=yes
extensions.conf:
[office]
exten => 11, 1, Dial(SIP/11,30)
;exten => 11, n, Playback(vmnobodyavail)
;exten => 11, n, Hangup()
exten => 12, 1, Dial(SIP/12,30)
;exten => 12, n, Playback(vmnobodyavail)
;exten => 12, n, Hangup()
exten => 13, 1, Dial(SIP/13,30)
;exten => 13, n, Playback(vmnobodyavail)
;exten => 13, n, Hangup()
exten => 14, 1, Dial(SIP/14,30)
;exten => 14, n, Playback(vmnobodyavail)
;exten => 14, n, Hangup()
;include => comtube_outbound
include => beeline_outbound
[comtube_outbound]
exten => _810.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _890.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _.{11},2,PlayBack(noanswer)
; exten => _.{11},4,HangUp
; exten => _.{11},5,PlayBack(busy)
; exten => _.{11},6,HangUp
[from_comtube]
exten => 775236,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 775236,2,Dial(SIP/11&SIP/12&SIP/13,25,Ttr) ; Входящие перенаправляются на внут$
exten => 775236,3,Hangup
[beeline_outbound]
exten => _.{6},1,Dial(SIP/beeline/${EXTEN},120)
;exten => _.{6},2,PlayBack(noanswer)
;exten => _.{6},4,HangUp
;exten => _.{6},5,PlayBack(busy)
;exten => _.{6},6,HangUp
[from_beeline]
exten => 313029,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 313029,2,Dial(SIP/11&SIP/12&SIP/13&SIP/14,25,Ttr) ; Входящие перенаправляются на внут$
exten => 313029,3,Hangup
Помогите понять, где я накосячил.
| | 3 | дополнение информации редактировать |
Debian 7.1, Asterisk 1.6.0.
SIP-транк от билайна включен в отдельную сетевуху eth1, статический ип.
ип. Добавлен статический маршрут по инструкции саппорта.
SIP-транк comtube идет через роутер, раздающий на офисную локалку интернет от билайна, тоже статический ип. Comtube используется только для звонков в европу, на номера, начинающиеся с 810.
Периодически на билайне пропадает слышимость собеседника, т.е. не слышно собеседника. Непонятно в связи с чем. На Comtube все ок, никаких проблем. externip=10.79.87.94 стоит ип сетевухи eth1.
Вот лог во время звонка на номер 060 (внешний):
<------------>
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Audio is at 10.79.87.94 port 14390
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x8 (alaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x4 (ulaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (NAT) to 10.225.61.130:5060:
INVITE sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
User-Agent: Topsail Asterisk PBX
Date: Tue, 16 Dec 2014 20:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 351041744 351041744 IN IP4 10.79.87.94
s=Asterisk PBX SVN-branch-1.6.0-r347660
c=IN IP4 10.79.87.94
t=0 0
m=audio 14390 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (6 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Got RTP packet from 11.12.13.5:10002 (type 08, seq 002308, ts 223115431, len 000160)
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
Reason: Q.850;cause=31;text="Normal, unspecified"
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Transmitting (NAT) to 10.225.61.130:5060:
ACK sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 ACK
User-Agent: Topsail Asterisk PBX
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Scheduling destruction of SIP dialog '139841780521131-221003236725599@11.12.13.5' in 32000 ms (Method: ACK)
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: Parsing <sip:11@11.12.13.5:5060> for address/port to send to
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: set destination to 11.12.13.5, port 5060
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (no NAT) to 11.12.13.5:5060:
BYE sip:11@11.12.13.5:5060 SIP/2.0
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
Max-Forwards: 70
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
User-Agent: Topsail Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
Server: Voip Phone 1.0
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: SIP Response message for INCOMING dialog BYE arrived
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '0ec920f346173cda626bbcc961afde88@10.225.61.130' Method: INVITE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '289305030e5e402416850bc957a1ae23@10.225.61.130' Method: BYE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '139841780521131-221003236725599@11.12.13.5' Method: ACK
[Dec 16 23:46:41] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.7:5060 --->
alive
<------------->
Постоянно идет инвайт на непонятный номер h.h. Уже давно, обратили внимание только сейчас, когда стала пропадать слышимость.
sip.conf:
register => ид:пароль@sip.comtube.com/ид
;register => 313029@10.225.61.130/313029
[beeline]
insecure=Invite,port
type=friend
fromuser=313029
canreinvite=no
context=from_beeline
qualify=yes
host=10.225.61.130
trunkname=beeline
dtmfmode=auto
nat=yes
srvlookup=yes
fromdomain=10.225.61.130
hassip=yes
dtmfmode=auto
[comtube]
username=хххххх
type=friend
secret=пароль
nat=yes
insecure=Invite,port
context=from_comtube
host=sip.comtube.com
trunkname=comtube
hassip=yes
fromuser=хххххх
fromdomain=sip.comtube.com
dtmfmode=auto
canreinvite=no
qualify=yes
extensions.conf:
[office]
exten => 11, 1, Dial(SIP/11,30)
;exten => 11, n, Playback(vmnobodyavail)
;exten => 11, n, Hangup()
exten => 12, 1, Dial(SIP/12,30)
;exten => 12, n, Playback(vmnobodyavail)
;exten => 12, n, Hangup()
exten => 13, 1, Dial(SIP/13,30)
;exten => 13, n, Playback(vmnobodyavail)
;exten => 13, n, Hangup()
exten => 14, 1, Dial(SIP/14,30)
;exten => 14, n, Playback(vmnobodyavail)
;exten => 14, n, Hangup()
;include => comtube_outbound
include => beeline_outbound
[comtube_outbound]
exten => _810.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _890.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _.{11},2,PlayBack(noanswer)
; exten => _.{11},4,HangUp
; exten => _.{11},5,PlayBack(busy)
; exten => _.{11},6,HangUp
[from_comtube]
exten => 775236,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 775236,2,Dial(SIP/11&SIP/12&SIP/13,25,Ttr) ; Входящие перенаправляются на внут$
exten => 775236,3,Hangup
[beeline_outbound]
exten => _.{6},1,Dial(SIP/beeline/${EXTEN},120)
;exten => _.{6},2,PlayBack(noanswer)
;exten => _.{6},4,HangUp
;exten => _.{6},5,PlayBack(busy)
;exten => _.{6},6,HangUp
[from_beeline]
exten => 313029,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 313029,2,Dial(SIP/11&SIP/12&SIP/13&SIP/14,25,Ttr) ; Входящие перенаправляются на внут$
exten => 313029,3,Hangup
Помогите понять, где я накосячил.накосячил, первый опыт.
Уточню если что нужно.
| | 4 | No.4 Revision редактировать |
Debian 7.1, Asterisk 1.6.0. SIP-транк от билайна включен в отдельную сетевуху eth1, статический ип. Добавлен статический маршрут по инструкции саппорта. SIP-транк comtube идет через роутер, раздающий на офисную локалку интернет от билайна, тоже статический ип. Comtube используется только для звонков в европу, на номера, начинающиеся с 810.
Периодически на билайне пропадает слышимость собеседника, т.е. не слышно собеседника. Непонятно в связи с чем. На Comtube все ок, никаких проблем. externip=10.79.87.94 стоит ип сетевухи eth1.
Вот лог во время звонка на номер 060 (внешний):
<------------>
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Audio is at 10.79.87.94 port 14390
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x8 (alaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding codec 0x4 (ulaw) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (NAT) to 10.225.61.130:5060:
INVITE sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
User-Agent: Topsail Asterisk PBX
Date: Tue, 16 Dec 2014 20:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 351041744 351041744 IN IP4 10.79.87.94
s=Asterisk PBX SVN-branch-1.6.0-r347660
c=IN IP4 10.79.87.94
t=0 0
m=audio 14390 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (6 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Got RTP packet from 11.12.13.5:10002 (type 08, seq 002308, ts 223115431, len 000160)
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://10.225.61.130:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.79.87.94:5060;received=10.79.87.94;branch=z9hG4bK4ffd77e1;rport=5060
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 INVITE
Reason: Q.850;cause=31;text="Normal, unspecified"
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Transmitting (NAT) to 10.225.61.130:5060:
ACK sip:h@10.225.61.130 SIP/2.0
Via: SIP/2.0/UDP 10.79.87.94:5060;branch=z9hG4bK4ffd77e1;rport
Max-Forwards: 70
From: "11" <sip:313029@10.225.61.130>;tag=as0097033b
To: <sip:h@10.225.61.130>;tag=SDvnsbf99-k45ax0v273
Contact: <sip:313029@10.79.87.94>
Call-ID: 0ec920f346173cda626bbcc961afde88@10.225.61.130
CSeq: 102 ACK
User-Agent: Topsail Asterisk PBX
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Scheduling destruction of SIP dialog '139841780521131-221003236725599@11.12.13.5' in 32000 ms (Method: ACK)
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: Parsing <sip:11@11.12.13.5:5060> for address/port to send to
[Dec 16 23:46:38] VERBOSE[2777] logger.c: set_destination: set destination to 11.12.13.5, port 5060
[Dec 16 23:46:38] VERBOSE[2777] logger.c: Reliably Transmitting (no NAT) to 11.12.13.5:5060:
BYE sip:11@11.12.13.5:5060 SIP/2.0
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
Max-Forwards: 70
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
User-Agent: Topsail Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Dec 16 23:46:38] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.12.13.2:5060;branch=z9hG4bK1f6562fd;rport
From: "060" <sip:060@11.12.13.2:5060>;tag=as6d9b5fe4
To: 11 <sip:11@11.12.13.2:5060>;tag=1070510280
Call-ID: 139841780521131-221003236725599@11.12.13.5
CSeq: 102 BYE
Server: Voip Phone 1.0
Content-Length: 0
<------------->
[Dec 16 23:46:38] VERBOSE[2507] logger.c: --- (8 headers 0 lines) ---
[Dec 16 23:46:38] VERBOSE[2507] logger.c: SIP Response message for INCOMING dialog BYE arrived
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '0ec920f346173cda626bbcc961afde88@10.225.61.130' Method: INVITE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '289305030e5e402416850bc957a1ae23@10.225.61.130' Method: BYE
[Dec 16 23:46:38] VERBOSE[2507] logger.c: Really destroying SIP dialog '139841780521131-221003236725599@11.12.13.5' Method: ACK
[Dec 16 23:46:41] VERBOSE[2507] logger.c:
<--- SIP read from UDP://11.12.13.7:5060 --->
alive
<------------->
Постоянно идет инвайт на непонятный номер h. Уже давно, обратили внимание только сейчас, когда стала пропадать слышимость.
sip.conf:
register => ид:пароль@sip.comtube.com/ид
;register => 313029@10.225.61.130/313029
[beeline]
insecure=Invite,port
type=friend
fromuser=313029
canreinvite=no
context=from_beeline
qualify=yes
host=10.225.61.130
trunkname=beeline
dtmfmode=auto
nat=yes
srvlookup=yes
fromdomain=10.225.61.130
hassip=yes
dtmfmode=auto
[comtube]
username=хххххх
type=friend
secret=пароль
nat=yes
insecure=Invite,port
context=from_comtube
host=sip.comtube.com
trunkname=comtube
hassip=yes
fromuser=хххххх
fromdomain=sip.comtube.com
dtmfmode=auto
canreinvite=no
qualify=yes
extensions.conf:
[office]
exten => 11, 1, Dial(SIP/11,30)
;exten => 11, n, Playback(vmnobodyavail)
;exten => 11, n, Hangup()
exten => 12, 1, Dial(SIP/12,30)
;exten => 12, n, Playback(vmnobodyavail)
;exten => 12, n, Hangup()
exten => 13, 1, Dial(SIP/13,30)
;exten => 13, n, Playback(vmnobodyavail)
;exten => 13, n, Hangup()
exten => 14, 1, Dial(SIP/14,30)
;exten => 14, n, Playback(vmnobodyavail)
;exten => 14, n, Hangup()
;include => comtube_outbound
include => beeline_outbound
[comtube_outbound]
exten => _810.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _890.,1,Dial(SIP/comtube/${EXTEN},120)
; exten => _.{11},2,PlayBack(noanswer)
; exten => _.{11},4,HangUp
; exten => _.{11},5,PlayBack(busy)
; exten => _.{11},6,HangUp
[from_comtube]
exten => 775236,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 775236,2,Dial(SIP/11&SIP/12&SIP/13,25,Ttr) ; Входящие перенаправляются на внут$
exten => 775236,3,Hangup
[beeline_outbound]
exten => _.{6},1,Dial(SIP/beeline/${EXTEN},120)
;exten => _.{6},2,PlayBack(noanswer)
;exten => _.{6},4,HangUp
;exten => _.{6},5,PlayBack(busy)
;exten => _.{6},6,HangUp
[from_beeline]
exten => 313029,1,Answer ; Входящие вызовы приходят на XXXXXX
exten => 313029,2,Dial(SIP/11&SIP/12&SIP/13&SIP/14,25,Ttr) ; Входящие перенаправляются на внут$
exten => 313029,3,Hangup
Помогите понять, где я накосячил, первый опыт. Уточню если что нужно.
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.