Code:
<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:06731705xx@192.168.1.10>
Content-Length: 0
<------------>
Audio is at 192.168.1.10 port 13190
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:06731705xx@192.168.1.10>
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 152311073 152311073 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.9-2
c=IN IP4 192.168.1.10
t=0 0
m=audio 13190 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
<------------>
<--- SIP read from UDP:192.168.1.20:5060 --->
CANCEL sip:06731705xx@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;rport
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone
Call-ID: 3641468165@192_168_1_20
CSeq: 3 CANCEL
Contact: <sip:kalidev1@192.168.1.20:5060>
Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:06731705xx@192.168.1.10", nonce="17667a62", response="41d4678f083049f066476f80d4b380e8"
Max-Forwards: 70
User-Agent: A580 IP021920000000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.20 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 CANCEL
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.20:5060 --->
ACK sip:06731705xx@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;rport
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 ACK
Contact: <sip:kalidev1@192.168.1.20:5060>
Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:06731705xx@192.168.1.10", nonce="17667a62", response="fdea256982ac7f340489f95dfba9f231"
Max-Forwards: 70
User-Agent: A580 IP021920000000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3641468165@192_168_1_20' Method: ACK
[Mar 22 10:47:21] NOTICE[8131]: chan_sip.c:11528 sip_reregister: -- Re-registration for 0033535541287@sip.ovh.net
[Mar 22 10:47:21] NOTICE[8131]: chan_sip.c:18270 handle_response_register: Outbound Registration: Expiry for sip.ovh.net is 120 sec (Scheduling reregistration in 105 s)
prodserver*CLI>
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