 | 1 | изначальная версия редактировать | |
Всем привет. Не смог найти аналогичную проблему среди заданных вопросов. В общем суть такая. Freepbx работает с двумя сетевыми картами. Добился успешной регистрации транка при помощи добавления маршрута. Входящие заработали. Приветствие, перевод звонка на внутреннего через IVR работают. Голос при входящих звонках отличный, с входящими проблем нет. Проблема с исходящими звонками - ну ни в какую не пойму из-за чего не идут звонки в мир. Подскажите пожалуйста в чем может быть такая проблема? Звоню с внутреннего номера 101. Заранее благодарю за подсказки и помощь.
username=1345068622
type=friend
secret=1234567890
nat=no
insecure=very
host=sip.telecom.kz
fromuser=1345068622
fromdomain=sip.telecom.kz
outboundproxy=10.0.0.12
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&alaw&g729
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7628a0d2",uri="sip:192.168.1.110",algorithm=MD5,response="0c17a75dfbf8a951296207c483dad49e"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cc74b9e"
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7cc74b9e",uri="sip:192.168.1.110",algorithm=MD5,response="df0f0a341a34dce628d103874c3d63c2"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
OPTIONS sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.9.0)
Date: Mon, 12 Nov 2012 10:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:101@192.168.1.50:5060>;expires=60
Date: Mon, 12 Nov 2012 10:26:05 GMT
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' in 6400 ms (Method: NOTIFY)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
NOTIFY sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(10.9.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.110
Voice-Message: 0/0 (0/0)
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 486 Busy Here
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060' Method: OPTIONS
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (8 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' Method: NOTIFY
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (15 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b482496"
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' in 6400 ms (Method: INVITE)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (16 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP TOS bits 184
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP CoS mark 5
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMA for ID 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMU for ID 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G723 for ID 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G726-32 for ID 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G729a for ID 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-40 for ID 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-24 for ID 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-16 for ID 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format NSE for ID 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Peer audio RTP is at port 192.168.1.50:16416
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Looking for 87051805545 in from-internal (domain 192.168.1.110)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: list_route: hop: <sip:101@192.168.1.50:5060>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:1] ResetCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:2] NoCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:3] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Audio is at 12148
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 142042388 142042388 IN IP4 192.168.1.110
s=Asterisk PBX 10.9.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 12148 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=ptime:20
a=sendrecv
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:4] Wait("SIP/101-00000080", "1") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:5] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:6] Playback("SIP/101-00000080", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'silence/1.alaw' (language 'en')
[2012-11-12 16:26:13] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
CANCEL sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="a13a653d57c150008206889c6c9a794b"
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, 87051805545, 6) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (11 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' Method: ACK
[2012-11-12 16:26:16] NOTICE[8586] manager.c: Seems to have passed...
| | 2 | No.2 Revision редактировать |
Всем привет. Не смог найти аналогичную проблему среди заданных вопросов. В общем суть такая. Freepbx работает с двумя сетевыми картами. Добился успешной регистрации транка при помощи добавления маршрута. Входящие заработали. Приветствие, перевод звонка на внутреннего через IVR работают. Голос при входящих звонках отличный, с входящими проблем нет. Проблема с исходящими звонками - ну ни в какую не пойму из-за чего не идут звонки в мир. Подскажите пожалуйста в чем может быть такая проблема? Звоню с внутреннего номера 101. Заранее благодарю за подсказки и помощь.
UPDATE: Это конечно смешно, но я решил проблему - создал исходящие правила. Я почему-то думал, что, если я совсем не буду создавать никаких исходящих правил, то выход в город на единственный транк будет производиться без каких-либо префиксов и т.д. Ну вот не хотелось набирать всякие девятки и т.д.:) Оказывается нет, к сожалению без исходящих правил никак. Странно только,что Астериск в логах показывал совсем не то. 401 Unauthorized - как то не серьезно. Я ожидал что-то вроде "outgoing rule not found"...а тут такие дела.:(
username=1345068622
type=friend
secret=1234567890
nat=no
insecure=very
host=sip.telecom.kz
fromuser=1345068622
fromdomain=sip.telecom.kz
outboundproxy=10.0.0.12
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&alaw&g729
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7628a0d2",uri="sip:192.168.1.110",algorithm=MD5,response="0c17a75dfbf8a951296207c483dad49e"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cc74b9e"
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7cc74b9e",uri="sip:192.168.1.110",algorithm=MD5,response="df0f0a341a34dce628d103874c3d63c2"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
OPTIONS sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.9.0)
Date: Mon, 12 Nov 2012 10:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:101@192.168.1.50:5060>;expires=60
Date: Mon, 12 Nov 2012 10:26:05 GMT
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' in 6400 ms (Method: NOTIFY)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
NOTIFY sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(10.9.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.110
Voice-Message: 0/0 (0/0)
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 486 Busy Here
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060' Method: OPTIONS
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (8 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' Method: NOTIFY
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (15 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b482496"
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' in 6400 ms (Method: INVITE)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (16 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP TOS bits 184
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP CoS mark 5
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMA for ID 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMU for ID 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G723 for ID 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G726-32 for ID 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G729a for ID 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-40 for ID 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-24 for ID 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-16 for ID 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format NSE for ID 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Peer audio RTP is at port 192.168.1.50:16416
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Looking for 87051805545 in from-internal (domain 192.168.1.110)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: list_route: hop: <sip:101@192.168.1.50:5060>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:1] ResetCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:2] NoCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:3] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Audio is at 12148
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 142042388 142042388 IN IP4 192.168.1.110
s=Asterisk PBX 10.9.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 12148 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=ptime:20
a=sendrecv
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:4] Wait("SIP/101-00000080", "1") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:5] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:6] Playback("SIP/101-00000080", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'silence/1.alaw' (language 'en')
[2012-11-12 16:26:13] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
CANCEL sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="a13a653d57c150008206889c6c9a794b"
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, 87051805545, 6) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (11 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' Method: ACK
[2012-11-12 16:26:16] NOTICE[8586] manager.c: Seems to have passed...
| | 3 | No.3 Revision редактировать |
Всем привет. Не смог найти аналогичную проблему среди заданных вопросов. В общем суть такая. Freepbx работает с двумя сетевыми картами. Добился успешной регистрации транка при помощи добавления маршрута. Входящие заработали. Приветствие, перевод звонка на внутреннего через IVR работают. Голос при входящих звонках отличный, с входящими проблем нет. Проблема с исходящими звонками - ну ни в какую не пойму из-за чего не идут звонки в мир. Подскажите пожалуйста в чем может быть такая проблема? Звоню с внутреннего номера 101. Заранее благодарю за подсказки и помощь.
UPDATE: Это конечно смешно, но я решил проблему - создал исходящие правила.
Я почему-то думал, что, если я совсем не буду создавать никаких исходящих правил, то выход в город на единственный транк будет производиться без каких-либо префиксов и т.д. Ну вот не хотелось набирать всякие девятки и т.д.:) Оказывается нет, к сожалению без исходящих правил никак.
Странно только,что Астериск в логах показывал совсем не то. 401 Unauthorized - как то не серьезно. Я ожидал что-то вроде "outgoing rule not found"...а тут такие дела.:(
username=1345068622
type=friend
secret=1234567890
nat=no
insecure=very
host=sip.telecom.kz
fromuser=1345068622
fromdomain=sip.telecom.kz
outboundproxy=10.0.0.12
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&alaw&g729
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7628a0d2",uri="sip:192.168.1.110",algorithm=MD5,response="0c17a75dfbf8a951296207c483dad49e"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cc74b9e"
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7cc74b9e",uri="sip:192.168.1.110",algorithm=MD5,response="df0f0a341a34dce628d103874c3d63c2"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
OPTIONS sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.9.0)
Date: Mon, 12 Nov 2012 10:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:101@192.168.1.50:5060>;expires=60
Date: Mon, 12 Nov 2012 10:26:05 GMT
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' in 6400 ms (Method: NOTIFY)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
NOTIFY sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(10.9.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.110
Voice-Message: 0/0 (0/0)
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 486 Busy Here
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060' Method: OPTIONS
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (8 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' Method: NOTIFY
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (15 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b482496"
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' in 6400 ms (Method: INVITE)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (16 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP TOS bits 184
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP CoS mark 5
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMA for ID 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMU for ID 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G723 for ID 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G726-32 for ID 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G729a for ID 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-40 for ID 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-24 for ID 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-16 for ID 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format NSE for ID 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Peer audio RTP is at port 192.168.1.50:16416
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Looking for 87051805545 in from-internal (domain 192.168.1.110)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: list_route: hop: <sip:101@192.168.1.50:5060>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:1] ResetCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:2] NoCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:3] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Audio is at 12148
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 142042388 142042388 IN IP4 192.168.1.110
s=Asterisk PBX 10.9.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 12148 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=ptime:20
a=sendrecv
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:4] Wait("SIP/101-00000080", "1") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:5] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:6] Playback("SIP/101-00000080", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'silence/1.alaw' (language 'en')
[2012-11-12 16:26:13] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
CANCEL sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="a13a653d57c150008206889c6c9a794b"
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, 87051805545, 6) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (11 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' Method: ACK
[2012-11-12 16:26:16] NOTICE[8586] manager.c: Seems to have passed...
| | 4 | No.4 Revision редактировать |
Всем привет. Не смог найти аналогичную проблему среди заданных вопросов. В общем суть такая. Freepbx работает с двумя сетевыми картами. Добился успешной регистрации транка при помощи добавления маршрута. Входящие заработали. Приветствие, перевод звонка на внутреннего через IVR работают. Голос при входящих звонках отличный, с входящими проблем нет. Проблема с исходящими звонками - ну ни в какую не пойму из-за чего не идут звонки в мир. Подскажите пожалуйста в чем может быть такая проблема? Звоню с внутреннего номера 101. Заранее благодарю за подсказки и помощь.
UPDATE: Это конечно смешно, но я решил проблему - создал исходящие правила. Я почему-то думал, что, если совсем не буду создавать никаких исходящих правил, то выход в город на единственный транк будет производиться без каких-либо префиксов и т.д. Ну вот не хотелось набирать всякие девятки и т.д.:) Оказывается нет, к сожалению без исходящих правил никак. Странно только,что Астериск в логах показывал совсем не то. 401 Unauthorized - как то не серьезно. Я ожидал что-то вроде "outgoing rule not found"...а тут такие дела.:(
username=1345068622
type=friend
secret=1234567890
nat=no
insecure=very
host=sip.telecom.kz
fromuser=1345068622
fromdomain=sip.telecom.kz
outboundproxy=10.0.0.12
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&alaw&g729
----------
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7628a0d2",uri="sip:192.168.1.110",algorithm=MD5,response="0c17a75dfbf8a951296207c483dad49e"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cc74b9e"
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7cc74b9e",uri="sip:192.168.1.110",algorithm=MD5,response="df0f0a341a34dce628d103874c3d63c2"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
OPTIONS sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.9.0)
Date: Mon, 12 Nov 2012 10:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:101@192.168.1.50:5060>;expires=60
Date: Mon, 12 Nov 2012 10:26:05 GMT
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' in 6400 ms (Method: NOTIFY)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
NOTIFY sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(10.9.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.110
Voice-Message: 0/0 (0/0)
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 486 Busy Here
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060' Method: OPTIONS
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (8 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' Method: NOTIFY
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (15 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b482496"
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' in 6400 ms (Method: INVITE)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (16 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP TOS bits 184
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP CoS mark 5
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMA for ID 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMU for ID 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G723 for ID 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G726-32 for ID 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G729a for ID 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-40 for ID 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-24 for ID 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-16 for ID 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format NSE for ID 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Peer audio RTP is at port 192.168.1.50:16416
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Looking for 87051805545 in from-internal (domain 192.168.1.110)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: list_route: hop: <sip:101@192.168.1.50:5060>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:1] ResetCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:2] NoCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:3] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Audio is at 12148
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 142042388 142042388 IN IP4 192.168.1.110
s=Asterisk PBX 10.9.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 12148 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=ptime:20
a=sendrecv
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:4] Wait("SIP/101-00000080", "1") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:5] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:6] Playback("SIP/101-00000080", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'silence/1.alaw' (language 'en')
[2012-11-12 16:26:13] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
CANCEL sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="a13a653d57c150008206889c6c9a794b"
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, 87051805545, 6) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (11 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' Method: ACK
[2012-11-12 16:26:16] NOTICE[8586] manager.c: Seems to have passed...
| | 5 | No.5 Revision редактировать |
Всем привет. Не смог найти аналогичную проблему среди заданных вопросов. В общем суть такая. Freepbx работает с двумя сетевыми картами. Добился успешной регистрации транка при помощи добавления маршрута. Входящие заработали. Приветствие, перевод звонка на внутреннего через IVR работают. Голос при входящих звонках отличный, с входящими проблем нет. Проблема с исходящими звонками - ну ни в какую не пойму из-за чего не идут звонки в мир. Подскажите пожалуйста в чем может быть такая проблема? Звоню с внутреннего номера 101. Заранее благодарю за подсказки и помощь.
UPDATE: Это конечно смешно, но я решил проблему - создал исходящие правила.
Я почему-то думал, что, если совсем не буду создавать никаких исходящих правил, то выход в город на единственный транк будет производиться без каких-либо префиксов и т.д. Ну вот не хотелось набирать всякие девятки и т.д.:) Оказывается нет, к сожалению без исходящих правил никак.
Странно только,что Астериск в логах показывал совсем не то. 401 Unauthorized - как то не серьезно. Я ожидал что-то вроде "outgoing rule not found"...а тут такие дела.:(дела.:(
Всем спасибо, тему закрываю.
username=1345068622
type=friend
secret=1234567890
nat=no
insecure=very
host=sip.telecom.kz
fromuser=1345068622
fromdomain=sip.telecom.kz
outboundproxy=10.0.0.12
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&alaw&g729
----------
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7628a0d2",uri="sip:192.168.1.110",algorithm=MD5,response="0c17a75dfbf8a951296207c483dad49e"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-675c1ac2;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23974 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cc74b9e"
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
REGISTER sip:192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="7cc74b9e",uri="sip:192.168.1.110",algorithm=MD5,response="df0f0a341a34dce628d103874c3d63c2"
Contact: "101" <sip:101@192.168.1.50:5060>;expires=60
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (13 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
OPTIONS sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.9.0)
Date: Mon, 12 Nov 2012 10:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-6daa41ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=ac6bfce17d79f505o0
To: "101" <sip:101@192.168.1.110>;tag=as213186ec
Call-ID: e7b1db0-55753110@192.168.1.50
CSeq: 23975 REGISTER
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:101@192.168.1.50:5060>;expires=60
Date: Mon, 12 Nov 2012 10:26:05 GMT
Content-Length: 0
<------------>
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' in 6400 ms (Method: NOTIFY)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.50:5060:
NOTIFY sip:101@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
To: <sip:101@192.168.1.50:5060>
Contact: <sip:Unknown@192.168.1.110:5060>
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(10.9.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.110
Voice-Message: 0/0 (0/0)
---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'e7b1db0-55753110@192.168.1.50' in 32000 ms (Method: REGISTER)
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 486 Busy Here
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as2fa2d007
Call-ID: 5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK651e5bde
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '5ea4ba9b4d1c48026cb4c3f02c9522f2@192.168.1.110:5060' Method: OPTIONS
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.1.50:5060>;tag=84594f11e5494895i0
From: "Unknown" <sip:Unknown@192.168.1.110>;tag=as643b9d2a
Call-ID: 40312a8665a038a11987970d220f4455@192.168.1.110:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK48307d46
Server: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: --- (8 headers 0 lines) ---
[2012-11-12 16:26:05] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog '40312a8665a038a11987970d220f4455@192.168.1.110:5060' Method: NOTIFY
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (15 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b482496"
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Scheduling destruction of SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' in 6400 ms (Method: INVITE)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-43f397ab
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as1d2829c0
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
INVITE sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Remote-Party-ID: "101" <sip:101@192.168.1.110>;screen=yes;party=calling
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 350011 350011 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 16416 RTP/AVP 8 0 4 2 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: --- (16 headers 20 lines) ---
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Using INVITE request as basis request - f71bbc0b-3b21852b@192.168.1.50
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found peer '101' for '101' from 192.168.1.50:5060
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP TOS bits 184
[2012-11-12 16:26:11] VERBOSE[1679] netsock2.c: == Using SIP RTP CoS mark 5
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found RTP audio format 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMA for ID 8
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format PCMU for ID 0
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G723 for ID 4
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G726-32 for ID 2
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format G729a for ID 18
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-40 for ID 96
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-24 for ID 97
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format G726-16 for ID 98
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found unknown media description format NSE for ID 100
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Peer audio RTP is at port 192.168.1.50:16416
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: Looking for 87051805545 in from-internal (domain 192.168.1.110)
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c: list_route: hop: <sip:101@192.168.1.50:5060>
[2012-11-12 16:26:11] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Length: 0
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:1] ResetCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:2] NoCDR("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:3] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Audio is at 12148
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-11-12 16:26:11] VERBOSE[8583] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:87051805545@192.168.1.110:5060>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 142042388 142042388 IN IP4 192.168.1.110
s=Asterisk PBX 10.9.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 12148 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=ptime:20
a=sendrecv
<------------>
[2012-11-12 16:26:11] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:4] Wait("SIP/101-00000080", "1") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:5] Progress("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] pbx.c: -- Executing [87051805545@from-internal:6] Playback("SIP/101-00000080", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2012-11-12 16:26:12] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'silence/1.alaw' (language 'en')
[2012-11-12 16:26:13] VERBOSE[8583] file.c: -- <SIP/101-00000080> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
CANCEL sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="a13a653d57c150008206889c6c9a794b"
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (10 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Sending to 192.168.1.50:5060 (no NAT)
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 INVITE
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f;received=192.168.1.50
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 CANCEL
Server: FPBX-2.10.1(10.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, 87051805545, 6) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/101-00000080", "") in new stack
[2012-11-12 16:26:15] VERBOSE[8583] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000080'
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c:
<--- SIP read from UDP:192.168.1.50:5060 --->
ACK sip:87051805545@192.168.1.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-196e31f
From: "101" <sip:101@192.168.1.110>;tag=eea14e8b53d514abo0
To: <sip:87051805545@192.168.1.110>;tag=as69f19e96
Call-ID: f71bbc0b-3b21852b@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="101",realm="asterisk",nonce="5b482496",uri="sip:87051805545@192.168.1.110",algorithm=MD5,response="5137b9616c8708ef8d06aa549de61732"
Contact: "101" <sip:101@192.168.1.50:5060>
User-Agent: Linksys/SPA2102-5.2.10
Content-Length: 0
<------------->
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: --- (11 headers 0 lines) ---
[2012-11-12 16:26:15] VERBOSE[1679] chan_sip.c: Really destroying SIP dialog 'f71bbc0b-3b21852b@192.168.1.50' Method: ACK
[2012-11-12 16:26:16] NOTICE[8586] manager.c: Seems to have passed...
Здесь был лог...
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.