<--- SIP read from 192.168.43.185:5060 --->
INVITE sip:81@192.168.43.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5c5f21cdfefa13e3
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>
Contact: <sip:1@192.168.43.185:5060>
Supported: replaces, timer
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58744 INVITE
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 253

v=0
o=1 8000 8000 IN IP4 192.168.43.185
s=SIP Call
c=IN IP4 192.168.43.185
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20

<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.43.185 : 5060 (no NAT)
Using INVITE request as basis request - 4d9f6a806ee19ba2@192.168.43.185

<--- Reliably Transmitting (no NAT) to 192.168.43.185:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5c5f21cdfefa13e3;received=192.168.43.185
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>;tag=as3a7d9cf5
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58744 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="705903c4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4d9f6a806ee19ba2@192.168.43.185' in 32000 ms (Method: INVITE)
Found user '1'
darkstar*CLI> 
<--- SIP read from 192.168.43.185:5060 --->
ACK sip:81@192.168.43.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5c5f21cdfefa13e3
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>;tag=as3a7d9cf5
Contact: <sip:1@192.168.43.185:5060>
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58744 ACK
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0    


<------------->
--- (11 headers 0 lines) ---
darkstar*CLI> 
<--- SIP read from 192.168.43.185:5060 --->
INVITE sip:81@192.168.43.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5ebbc3e0a5932c1e
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>
Contact: <sip:1@192.168.43.185:5060>
Supported: replaces, timer
Proxy-Authorization: Digest username="1", realm="asterisk", algorithm=MD5, uri="sip:81@192.168.43.157", nonce="705903c4", response="9f4ff7702242cc554a7add69387d97a3"
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58745 INVITE
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 253

v=0
o=1 8000 8001 IN IP4 192.168.43.185
s=SIP Call
c=IN IP4 192.168.43.185
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20

<------------->
--- (14 headers 13 lines) ---
Sending to 192.168.43.185 : 5060 (no NAT)
Using INVITE request as basis request - 4d9f6a806ee19ba2@192.168.43.185
Found user '1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Peer audio RTP is at port 192.168.43.185:5004
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G723 for ID 4
Found description format G729 for ID 18
Found description format GSM for ID 3
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.43.185:5004
Looking for 81 in internal (domain 192.168.43.157)
list_route: hop: <sip:1@192.168.43.185:5060>

<--- Transmitting (no NAT) to 192.168.43.185:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5ebbc3e0a5932c1e;received=192.168.43.185
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58745 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:81@192.168.43.157>
Content-Length: 0


<------------>
Audio is at 192.168.43.157 port 25554
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.43.158:5060:
INVITE sip:1@192.168.43.158 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.157:5060;branch=z9hG4bK6c4e9537;rport
From: "Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a
To: <sip:1@192.168.43.158>
Contact: <sip:1@192.168.43.157>
Call-ID: 37cc53584b23f55566aa7c162cc8794e@192.168.43.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Aug 2007 07:35:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 5204 5204 IN IP4 192.168.43.157
s=session
c=IN IP4 192.168.43.157
t=0 0
m=audio 25554 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

---
darkstar*CLI> 
<--- SIP read from 192.168.43.158:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.43.157:5060;branch=z9hG4bK6c4e9537;received=192.168.43.157;rport=5060
From: "Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a
To: <sip:1@192.168.43.158>;tag=as4e9bd32e
Call-ID: 37cc53584b23f55566aa7c162cc8794e@192.168.43.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c2ebc2f"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 192.168.43.158:5060:
ACK sip:1@192.168.43.158 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.157:5060;branch=z9hG4bK6c4e9537;rport
From: "Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a
To: <sip:1@192.168.43.158>;tag=as4e9bd32e
Contact: <sip:1@192.168.43.157>
Call-ID: 37cc53584b23f55566aa7c162cc8794e@192.168.43.157
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 192.168.43.157 port 25554
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.43.158:5060:
INVITE sip:1@192.168.43.158 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.157:5060;branch=z9hG4bK70fb67ed;rport
From: "Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a
To: <sip:1@192.168.43.158>
Contact: <sip:1@192.168.43.157>
Call-ID: 37cc53584b23f55566aa7c162cc8794e@192.168.43.157
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="asterisk2", realm="asterisk", algorithm=MD5, uri="sip:1@192.168.43.158", nonce="6c2ebc2f", response="a20456dbdc726d5f2d271eff938aca66", opaque=""
Date: Tue, 28 Aug 2007 07:35:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 5204 5205 IN IP4 192.168.43.157
s=session
c=IN IP4 192.168.43.157
t=0 0
m=audio 25554 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

---
darkstar*CLI> 
<--- SIP read from 192.168.43.158:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.43.157:5060;branch=z9hG4bK70fb67ed;received=192.168.43.157;rport=5060
From: "Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a
To: <sip:1@192.168.43.158>;tag=as4e9bd32e
Call-ID: 37cc53584b23f55566aa7c162cc8794e@192.168.43.157
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.43.158:5060:
ACK sip:1@192.168.43.158 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.157:5060;branch=z9hG4bK70fb67ed;rport
From: "Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a
To: <sip:1@192.168.43.158>;tag=as4e9bd32e
Contact: <sip:1@192.168.43.157>
Call-ID: 37cc53584b23f55566aa7c162cc8794e@192.168.43.157
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Aug 28 17:05:27] WARNING[5212]: chan_sip.c:12016 handle_response_invite: Received response: "Forbidden" from '"Ast1Ext1" <sip:1@192.168.43.157>;tag=as1fd4323a'
darkstar*CLI> 
<--- Transmitting (no NAT) to 192.168.43.185:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5ebbc3e0a5932c1e;received=192.168.43.185
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>;tag=as4ff9ecd0
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58745 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:81@192.168.43.157>
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21


<------------>
darkstar*CLI> 
<--- SIP read from 192.168.43.185:5060 --->
ACK sip:81@192.168.43.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.185:5060;branch=z9hG4bK5ebbc3e0a5932c1e
From: "hugo testing" <sip:1@192.168.43.157>;tag=d8fc0fc0c14012b3
To: <sip:81@192.168.43.157>;tag=as4ff9ecd0
Contact: <sip:1@192.168.43.185:5060>
Proxy-Authorization: Digest username="1", realm="asterisk", algorithm=MD5, uri="sip:81@192.168.43.157", nonce="705903c4", response="a7291cb0aa6456e89027a471a2ecc488"
Call-ID: 4d9f6a806ee19ba2@192.168.43.185
CSeq: 58745 ACK
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0    


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '37cc53584b23f55566aa7c162cc8794e@192.168.43.157' Method: INVITE
Really destroying SIP dialog '4d9f6a806ee19ba2@192.168.43.185' Method: ACK