電訊茶室 - Powered by Discuz! Board

[general]
srvlookup = yes
realm=hostname_of_your_wan_port
externhost=hostname_of_your_wan_port
fromdomain=hostname_of_your_wan_port
address
localnet=10.0.88.0/255.255.255.0 ;change it as per your Asterisk network address
externrefresh = 1
defaultexpirey=360
bindport=5080
bindaddr=10.0.88.6
nat=force_rport,comedia
qualify=yes
disallow=all
allow=ulaw,alaw,gsm
alwaysauthreject=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
pedantic=yes
context=front-desk
register => 333445566:password@ngn2.nwtbb.com/333445566
[nwt-nettalk]
username=333445566
type=peer
secret=password
qualify=yes
port=5060
nat=force_rport,comedia
insecure = port,invite
host=ngn2.nwtbb.com
fromusername=333445566
fromuser=333445566
fromdomain=ngn2.nwtbb.com
dtmfmode = rfc2833
canreinvite = no
defaultexpirey=300
context = from-nwt-nettalk
[2004]
type=friend
secret=password_for_2004
qualify=yes
host=dynamic
nat=force_rport,comedia
insecure=invite
canreinvit=no
context=internal

複製代碼

[front-desk]
[internal]
exten => _XXX.,1,Dial(SIP/${EXTEN}@nwt-nettalk,,r)
exten => 2004,1,Dial(SIP/2004,,r)
[from-nwt-nettalk]
exten => 33445566,1,Dial(SIP/2004)
exten => 33445566,2,Hangup

複製代碼

TWTS-269PRO*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [99663311@internal:1] Dial("SIP/2004-0000003c", "SIP/96331527@nwt-nettalk,,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/99663311@nwt-nettalk
[Aug 4 09:50:39] WARNING[15750][C-0000001f]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
[Aug 4 09:50:39] WARNING[15750][C-0000001f]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
-- SIP/nwt-nettalk-0000003d is ringing
-- SIP/nwt-nettalk-0000003d is making progress passing it to SIP/2004-0000003c
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>

複製代碼

TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
[Aug 4 09:50:47] WARNING[15750][C-0000001f]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
[Aug 4 09:50:47] WARNING[15750][C-0000001f]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
-- SIP/nwt-nettalk-0000003d answered SIP/2004-0000003c
== Spawn extension (internal, 99663311, 1) exited non-zero on 'SIP/2004-0000003c'
-- Got SIP response 500 "Server Internal Error" back from 203.176.254.198:5060
TWTS-269PRO*CLI>

複製代碼

TWTS-269PRO*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [33445566@from-nwt-nettalk:1] Dial("SIP/nwt-nettalk-0000003e", "SIP/2004") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/2004
-- SIP/2004-0000003f is ringing
..
..
我接电话后出现下面的信息：
..
..
-- SIP/2004-0000003f answered SIP/nwt-nettalk-0000003e
-- Locally bridging SIP/nwt-nettalk-0000003e and SIP/2004-0000003f
..
..
我说完收线
..
..
== Spawn extension (from-nwt-nettalk, 33445566, 1) exited non-zero on 'SIP/nwt-nettalk-0000003e'

複製代碼

TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
<--- SIP read from UDP:10.0.88.22:5060 --->
INVITE sip:99663311@10.0.88.6:5080 SIP/2.0
Via: SIP/2.0/UDP 10.0.88.22:5060;rport;branch=z9hG4bKff918c1e86
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
Contact: <sip:2004@10.0.88.22:5060>
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Supported: replaces
Content-Type: application/sdp
User-Agent: DGP306-O (1304100)
Content-Length: 217
v=0
o=CMI-SIPUA 477 0 IN IP4 10.0.88.22
s=SIP CALL
c=IN IP4 10.0.88.22
t=0 0
m=audio 21864 RTP/AVP 0 8 4 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=rtcp:21865
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to 10.0.88.22:5060 (NAT)
Sending to 10.0.88.22:5060 (NAT)
Using INVITE request as basis request - [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
Found peer '2004' for '2004' from 10.0.88.22:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.88.22:21864
Looking for 99663311 in internal (domain 10.0.88.6)
list_route: hop: <sip:2004@10.0.88.22:5060>
<--- Transmitting (NAT) to 10.0.88.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 1 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:99663311@10.0.88.6:5080>
Content-Length: 0
<------------>
-- Executing [99663311@internal:1] Dial("SIP/2004-00000042", "SIP/99663311@nwt-nettalk,,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 12134
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.176.254.198:5060:
INVITE sip:99663311@ngn2.nwtbb.com:5060 SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport
Max-Forwards: 70
From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>
Contact: <sip:33445566@219.73.68.130:5080>
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.0
Date: Sun, 04 Aug 2013 06:26:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 1811917192 1811917192 IN IP4 219.73.68.130
s=Asterisk PBX 11.5.0
c=IN IP4 219.73.68.130
t=0 0
m=audio 12134 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/99663311@nwt-nettalk
<--- Transmitting (NAT) to 10.0.88.22:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 1 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:99663311@10.0.88.6:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=hcxvyhoy
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="Huawei",nonce="14:25:51:59515",stale=false,algorithm=MD5
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.176.254.198:5060:
ACK sip:99663311@ngn2.nwtbb.com:5060 SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport
Max-Forwards: 70
From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=hcxvyhoy
Contact: <sip:33445566@219.73.68.130:5080>
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.5.0
Content-Length: 0
---
Audio is at 12134
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.176.254.198:5060:
INVITE sip:99663311@ngn2.nwtbb.com:5060 SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport
Max-Forwards: 70
From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>
Contact: <sip:33445566@219.73.68.130:5080>
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.5.0
Proxy-Authorization: Digest username="33445566", realm="Huawei", algorithm=MD5, uri="sip:99663311@ngn2.nwtbb.com:5060", nonce="14:25:51:59515", response="e7c53f4651865f857b5f4d567752819e"
Date: Sun, 04 Aug 2013 06:26:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 1811917192 1811917193 IN IP4 219.73.68.130
s=Asterisk PBX 11.5.0
c=IN IP4 219.73.68.130
t=0 0
m=audio 12134 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

複製代碼

<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>
CSeq: 103 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
CSeq: 103 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:203.176.254.198:5060;user=phone>
Content-Length: 301
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 30838616 30838616 IN IP4 203.176.254.198
s=Sip Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=silenceSupp:off - - - -
a=ecan:fb on -
a=X-fax
a=fmtp:101 0-15
a=inactive
<------------->
--- (10 headers 15 lines) ---
list_route: hop: <sip:203.176.254.198:5060;user=phone>
[Aug 4 14:26:18] WARNING[15750][C-00000022]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
[Aug 4 14:26:18] WARNING[15750][C-00000022]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
-- SIP/nwt-nettalk-00000043 is ringing
<--- Transmitting (NAT) to 10.0.88.22:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 1 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:99663311@10.0.88.6:5080>
Content-Length: 0
<------------>
-- SIP/nwt-nettalk-00000043 is making progress passing it to SIP/2004-00000042
Reliably Transmitting (NAT) to 203.176.254.198:5060:
OPTIONS sip:ngn2.nwtbb.com SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK5095bc48;rport
Max-Forwards: 70
From: "asterisk" <sip:33445566@vpntw.homeftp.org>;tag=as69c8458c
To: <sip:ngn2.nwtbb.com>
Contact: <sip:33445566@219.73.68.130:5080>
Call-ID: [email]46cfd6b96d486efd1ef09f185beb0fc2@vpntw.homeftp.org[/email]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.5.0
Date: Sun, 04 Aug 2013 06:26:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK5095bc48;rport=5080
Call-ID: [email]46cfd6b96d486efd1ef09f185beb0fc2@vpntw.homeftp.org[/email]
From: "asterisk"<sip:33445566@vpntw.homeftp.org>;tag=as69c8458c
To: <sip:ngn2.nwtbb.com>;tag=culhbyoh
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '46cfd6b96d486efd1ef09f185beb0fc2@vpntw.homeftp.org' Method: OPTIONS
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>

複製代碼

TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
TWTS-269PRO*CLI>
Really destroying SIP dialog '5f3cf05319c609fe18ba0fea5f6ad854@10.0.88.22' Method: REGISTER
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
CSeq: 103 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:203.176.254.198:5060;user=phone>
Content-Length: 301
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 30838616 30838617 IN IP4 203.176.254.198
s=Sip Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=silenceSupp:off - - - -
a=ecan:fb on -
a=X-fax
a=fmtp:101 0-15
a=inactive
<------------->
--- (10 headers 15 lines) ---
[Aug 4 14:26:33] WARNING[15750][C-00000022]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
[Aug 4 14:26:33] WARNING[15750][C-00000022]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
list_route: hop: <sip:203.176.254.198:5060;user=phone>
set_destination: Parsing <sip:203.176.254.198:5060;user=phone> for address/port to send to
set_destination: set destination to 203.176.254.198:5060
Transmitting (NAT) to 203.176.254.198:5060:
ACK sip:203.176.254.198:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK734c65bc;rport
Max-Forwards: 70
From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
Contact: <sip:33445566@219.73.68.130:5080>
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.5.0
Content-Length: 0
---
set_destination: Parsing <sip:203.176.254.198:5060;user=phone> for address/port to send to
set_destination: set destination to 203.176.254.198:5060
Reliably Transmitting (NAT) to 203.176.254.198:5060:
BYE sip:203.176.254.198:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK392c49ed;rport
Max-Forwards: 70
From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
CSeq: 104 BYE
User-Agent: Asterisk PBX 11.5.0
Proxy-Authorization: Digest username="33445566", realm="Huawei", algorithm=MD5, uri="sip:203.176.254.198:5060", nonce="14:25:51:59515", response="19d6ae396e735c889d4a12b4db6b92b8"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

複製代碼

---
Scheduling destruction of SIP dialog '2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com' in 6400 ms (Method: INVITE)
-- SIP/nwt-nettalk-00000043 answered SIP/2004-00000042
Audio is at 12990
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.0.88.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 1 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:99663311@10.0.88.6:5080>
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 837645094 837645094 IN IP4 10.0.88.6
s=Asterisk PBX 11.5.0
c=IN IP4 10.0.88.6
t=0 0
m=audio 12990 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Scheduling destruction of SIP dialog '2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:203.176.254.198:5060;user=phone> for address/port to send to
set_destination: set destination to 203.176.254.198:5060
Reliably Transmitting (NAT) to 203.176.254.198:5060:
BYE sip:203.176.254.198:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK20935ba0;rport
Max-Forwards: 70
From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
CSeq: 105 BYE
User-Agent: Asterisk PBX 11.5.0
Proxy-Authorization: Digest username="33445566", realm="Huawei", algorithm=MD5, uri="sip:203.176.254.198:5060", nonce="14:25:51:59515", response="19d6ae396e735c889d4a12b4db6b92b8"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (internal, 99663311, 1) exited non-zero on 'SIP/2004-00000042'
Scheduling destruction of SIP dialog '790da6076f0f53a7573d31e77297616f@10.0.88.22' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK20935ba0;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
CSeq: 105 BYE
Warning: 399 SE2000 "SSF00157L02501[5645] Unexpected message received"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 500 "Server Internal Error" back from 203.176.254.198:5060
<--- SIP read from UDP:203.176.254.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK392c49ed;rport=5080
Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
CSeq: 104 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com' Method: INVITE
Retransmitting #1 (NAT) to 10.0.88.22:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 1 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:99663311@10.0.88.6:5080>
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 837645094 837645094 IN IP4 10.0.88.6
s=Asterisk PBX 11.5.0
c=IN IP4 10.0.88.6
t=0 0
m=audio 12990 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:10.0.88.22:5060 --->
ACK sip:99663311@10.0.88.6:5080 SIP/2.0
Via: SIP/2.0/UDP 10.0.88.22:5060;rport;branch=z9hG4bKba10cf7ffe
From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
Contact: <sip:2004@10.0.88.22:5060>
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:2004@10.0.88.22:5060> for address/port to send to
set_destination: set destination to 10.0.88.22:5060
Reliably Transmitting (NAT) to 10.0.88.22:5060:
BYE sip:2004@10.0.88.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.88.6:5080;branch=z9hG4bK1bbfa85a;rport
Max-Forwards: 70
From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Scheduling destruction of SIP dialog '790da6076f0f53a7573d31e77297616f@10.0.88.22' in 6400 ms (Method: ACK)
Retransmitting #1 (NAT) to 10.0.88.22:5060:
BYE sip:2004@10.0.88.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.88.6:5080;branch=z9hG4bK1bbfa85a;rport
Max-Forwards: 70
From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.0.88.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.88.6:5080;rport=5080;received=10.0.88.6;branch=z9hG4bK1bbfa85a
From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '790da6076f0f53a7573d31e77297616f@10.0.88.22' Method: ACK
<--- SIP read from UDP:10.0.88.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.88.6:5080;rport=5080;received=10.0.88.6;branch=z9hG4bK1bbfa85a
From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
TWTS-269PRO*CLI>

複製代碼