INVITE sip:[EMAIL PROTECTED] SIP/2.0
l: 214
m: <sip:[EMAIL PROTECTED]:5060>
i: [EMAIL PROTECTED]
c: application/sdp
Max-Forwards: 70
CSeq: 13 INVITE
f: <sip:[EMAIL PROTECTED]:2841>;tag=41280171719448
t: <sip:[EMAIL PROTECTED]>;tag=as7cf27066
User-Agent: SJLabs-SJphone/1.30.252
v: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078
v=0
o=- 3318544820 3318544833 IN IP4
192.168.1.111
s=SJphone
c=IN IP4 0.0.0.0
t=0
0
a=direction:active
m=audio 16394 RTP/AVP 3 101
a=rtpmap:3
GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
11 headers, 10 lines
Using latest request as basis
request
Sending to 192.168.1.111 : 5060 (NAT)
Found audio format
UNKN
Found audio format UNKN
Found description format GSM
Found
description format telephone-event
Capabilities: us - 6, them - 2/0, combined
- 2
Non-codec capabilities: us - 1, them - 1, combined - 1
We're
at xxx.xxx.xxx.xxx port 14276
Answering with preferred capability
2
Answering with non-codec capability 1
Reliably Transmitting
(NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078;received=xxx.xxx.xxx.xxx
From:
<sip:[EMAIL PROTECTED]:2841>;tag=41280171719448
To:
<sip:[EMAIL PROTECTED]>;tag=as7cf27066
Call-ID: [EMAIL PROTECTED]
CSeq:
13 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type:
application/sdp
Content-Length: 219
v=0
o=root 17002 17015 IN IP4
xxx.xxx.xxx.xxx
s=session
c=IN IP4 195.216.65.216
t=0 0
m=audio
14276 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101
telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to xxx.xxx.xxx.xxx:2841
*********************** NOT WORKING ********************************
Sip read:
INVITE sip:[EMAIL PROTECTED]
SIP/2.0
l: 214
m: <sip:[EMAIL PROTECTED]:5060>
i: [EMAIL PROTECTED]
c:
application/sdp
Max-Forwards: 70
CSeq: 1 INVITE
f:
<sip:[EMAIL PROTECTED]:5060>;tag=41308811925234
t:
<sip:[EMAIL PROTECTED]>;tag=as463b04a6
User-Agent:
SJLabs-SJphone/1.30.252
v: SIP/2.0/UDP
192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8
v=0
o=- 3318545106 3318545107 IN IP4
192.168.1.111
s=SJphone
c=IN IP4 0.0.0.0
t=0
0
a=direction:active
m=audio 16400 RTP/AVP 3 101
a=rtpmap:3
GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
11 headers, 10 lines
Using latest request as basis
request
Sending to 192.168.1.111 : 5060 (NAT)
We're at 192.168.1.20 port
18336
Answering/Requesting with root capability 0x4
(ulaw)
Answering with preferred capability 0x2 (gsm)
Answering with
non-codec capability 0x1 (telephone-event)
Reliably Transmitting
(NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8;received=192.168.1.111;rport=5060
From:
<sip:[EMAIL PROTECTED]:5060>;tag=41308811925234
To:
<sip:[EMAIL PROTECTED]>;tag=as463b04a6
Call-ID: [EMAIL PROTECTED]
CSeq:
1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type:
application/sdp
Content-Length: 241
v=0
o=root 12791 12793 IN IP4
192.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 0
m=audio 16398
RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101
telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.1.111:5060
Thanks
KF
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