GV打出，平板接聽正常（800-555-8355 或其他800電話）。下麵是平板打出不通的trace report和Siporcery Deflaut dial plan，不知大俠能否望診一下。謝了！
SIPTransaction=> SIPTransaction=>Request received udp:69.59.142.213:5070<-udp:69
INVITE sip:18005558355@sipsorcery.com SIP/2.0
Via: SIP/2.0/UDP 69.59.142.213:5060;branch=
Via: SIP/2.0/UDP 77.102.154.183:40847;rport=
To: <sip:18005558355@sipsorcery.
From: <sip:373216@sipsorcery.
Call-ID: aNWidldKgT8MrYO4LLL.cyulY-uE.
CSeq: 27103 INVITE
Contact: <sip:373216@77.102.154.183:40847;transport=UDP;ob>
Max-Forwards: 69
Route: <sip:sipsorcery.com;transport=
User-Agent: CSipSimple r944 / harmony-8
Authorization: Digest username="373216",realm=
Supported: replaces, 100rel, timer, norefersub
Content-Length: 376
Content-Type: application/sdp
Proxy-ReceivedFrom: udp:77.102.154.183:40847
Proxy-ReceivedOn: udp:69.59.142.213:5060
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Session-Expires: 1800
Min-SE: 90
v=0
o=- 3522267487 3522267487 IN IP4 10.0.0.9
s=pjmedia
c=IN IP4 10.0.0.9
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 104 103 102 0 8 101
a=rtcp:4001 IN IP4 10.0.0.9
a=rtpmap:9 G722/8000
a=rtpmap:104 speex/32000
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
DialPlan=> Dialplan trace commenced at 13 Aug 2011 16:38:48:812.
DialPlan=> Log message from default dialplan.
DialPlan=> Commencing Dial with: music@iptel.org.
DialPlan=> ForkCall commencing call leg to sip:music@iptel.org.
DialPlan=> Switching to sip:music@iptel.org:5060 via udp:69.59.142.213:5060.
DialPlan=> SDP on UAC call had RTP socket mangled from 10.0.0.9:4000 to 77.102.154.183:4000.
SIPTransaction=> Send Request reliable udp:69.59.142.213:5070->udp:69
INVITE sip:music@iptel.org SIP/2.0
Via: SIP/2.0/UDP 69.59.142.213:5070;branch=
To: <sip:music@iptel.org>
From: <sip:373216@sipsorcery.
Call-ID: d4e350521c3c44d9a37fcd84620085
CSeq: 1 INVITE
Contact: <sip:69.59.142.213:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Length: 382
Content-Type: application/sdp
v=0
o=- 3522267487 3522267487 IN IP4 10.0.0.9
s=pjmedia
c=IN IP4 77.102.154.183
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 104 103 102 0 8 101
a=rtcp:4001 IN IP4 10.0.0.9
a=rtpmap:9 G722/8000
a=rtpmap:104 speex/32000
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIPTransaction=> Received Response udp:69.59.142.213:5070<-udp:69
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 69.59.142.213:5070;branch=
To: <sip:music@iptel.org>
From: <sip:373216@sipsorcery.
Call-ID: d4e350521c3c44d9a37fcd84620085
CSeq: 1 INVITE
Content-Length: 0
Proxy-ReceivedFrom: udp:213.192.59.75:5060
Proxy-ReceivedOn: udp:69.59.142.213:5060
Server: ser (3.2.0-dev2 (i386/linux))
Warning: 392 213.192.59.75:5060 "Noisy feedback tells: pid=18980 req_src_ip=69.59.142.213 req_src_port=5060 in_uri=sip:music@iptel.org out_uri=sip:music@iptel.org via_cnt==1"
DialPlan=> Information response 100 trying -- your call is important to us for sip:music@iptel.org.
SIPTransaction=> Received Response udp:69.59.142.213:5070<-udp:69
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.59.142.213:5070;branch=
To: <sip:music@iptel.org>;tag=
From: <sip:373216@sipsorcery.
Call-ID: d4e350521c3c44d9a37fcd84620085
CSeq: 1 INVITE
Contact: <sip:music@213.192.59.73:5080>
Record-Route: <sip:213.192.59.75;ftag=
Content-Length: 254
Content-Type: application/sdp
Proxy-ReceivedFrom: udp:213.192.59.75:5060
Proxy-ReceivedOn: udp:69.59.142.213:5060
Server: Sip Express Media Server (1.4.1 (x86/linux))
v=0
o=sems 1130205416 1934993233 IN IP4 213.192.59.73
s=session
c=IN IP4 213.192.59.73
t=0 0
m=audio 15054 RTP/AVP 102 0 8 101
a=rtpmap:102 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIPTransaction=> Send Request udp:69.59.142.213:5070->udp:69
ACK sip:music@213.192.59.73:5080 SIP/2.0
Via: SIP/2.0/UDP 69.59.142.213:5070;branch=
To: <sip:music@iptel.org>;tag=
From: <sip:373216@sipsorcery.
Call-ID: d4e350521c3c44d9a37fcd84620085
CSeq: 1 ACK
Max-Forwards: 70
Route: <sip:213.192.59.75;ftag=
Content-Length: 0
DialPlan=> Response 200 OK for sip:music@iptel.org.
DialPlan=> SDP on UAC response had public IP not mangled, RTP socket 213.192.59.73:15054.
DialPlan=> Cancelling all call legs for ForkCall app.
SIPTransaction=> Send Final Response Reliable udp:69.59.142.213:5070->69.59.
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 69.59.142.213:5060;branch=
Via: SIP/2.0/UDP 77.102.154.183:40847;rport=
To: <sip:18005558355@sipsorcery.
From: <sip:373216@sipsorcery.
Call-ID: aNWidldKgT8MrYO4LLL.cyulY-uE.
CSeq: 27103 INVITE
Contact: <sip:69.59.142.213:5070>
Server: www.sipsorcery.com
Content-Length: 254
Content-Type: application/sdp
v=0
o=sems 1130205416 1934993233 IN IP4 213.192.59.73
s=session
c=IN IP4 213.192.59.73
t=0 0
m=audio 15054 RTP/AVP 102 0 8 101
a=rtpmap:102 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
DialPlan=> Dial command was successfully answered in 0.27s.
DialPlan=> Dialplan cleanup for 373216.
DialPlan=> Dialplan trace completed at 13 Aug 2011 16:39:21:015.
============== Siporcery deflaut dial plan ==============
type: Ruby
sys.Log("Log message from default dialplan.")
sys.Dial("music@iptel.org")
sys.GoogleVoiceCall("373216@gmail.com","88888888","2538888888","#{ req.URI.User }",".*","3")
