rendered paste body<--- SIP read from 10.0.1.41:5060 --->
INVITE sip:3102@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4d779f52
From: <sip:3101@10.0.1.21>;tag=be9c545d28cb6c1ao0
To: <sip:3102@10.0.1.21>
Call-ID: a50a6fd9-a559a6ce@10.0.1.41
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:3101@10.0.1.41:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 387
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 19028 19028 IN IP4 10.0.1.41
s=-
c=IN IP4 10.0.1.41
t=0 0
m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 10.0.1.41 : 5060 (no NAT)
Using INVITE request as basis request - a50a6fd9-a559a6ce@10.0.1.41
<--- Reliably Transmitting (no NAT) to 10.0.1.41:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4d779f52;received=10.0.1.41
From: <sip:3101@10.0.1.21>;tag=be9c545d28cb6c1ao0
To: <sip:3102@10.0.1.21>;tag=as25711cfe
Call-ID: a50a6fd9-a559a6ce@10.0.1.41
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e7017fd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'a50a6fd9-a559a6ce@10.0.1.41' in 32000 ms (Method: INVITE)
Found user '3101'
akira*CLI>
<--- SIP read from 10.0.1.41:5060 --->
ACK sip:3102@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4d779f52
From: <sip:3101@10.0.1.21>;tag=be9c545d28cb6c1ao0
To: <sip:3102@10.0.1.21>;tag=as25711cfe
Call-ID: a50a6fd9-a559a6ce@10.0.1.41
CSeq: 101 ACK
Max-Forwards: 70
Contact: <sip:3101@10.0.1.41:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
akira*CLI>
<--- SIP read from 10.0.1.41:5060 --->
INVITE sip:3102@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-bfdeb27
From: <sip:3101@10.0.1.21>;tag=be9c545d28cb6c1ao0
To: <sip:3102@10.0.1.21>
Call-ID: a50a6fd9-a559a6ce@10.0.1.41
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="5e7017fd",uri="sip:3102@10.0.1.21",algorithm=MD5,response="f803cbb941ef4649a92794ca5cd36433"
Contact: <sip:3101@10.0.1.41:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 387
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 19028 19028 IN IP4 10.0.1.41
s=-
c=IN IP4 10.0.1.41
t=0 0
m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.0.1.41 : 5060 (no NAT)
Using INVITE request as basis request - a50a6fd9-a559a6ce@10.0.1.41
Found user '3101'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.0.1.41:16456
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.1.41:16456
Looking for 3102 in handsets (domain 10.0.1.21)
<--- Reliably Transmitting (no NAT) to 10.0.1.41:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-bfdeb27;received=10.0.1.41
From: <sip:3101@10.0.1.21>;tag=be9c545d28cb6c1ao0
To: <sip:3102@10.0.1.21>;tag=as25711cfe
Call-ID: a50a6fd9-a559a6ce@10.0.1.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'a50a6fd9-a559a6ce@10.0.1.41' in 32000 ms (Method: INVITE)
akira*CLI>
<--- SIP read from 10.0.1.41:5060 --->
ACK sip:3102@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-bfdeb27
From: <sip:3101@10.0.1.21>;tag=be9c545d28cb6c1ao0
To: <sip:3102@10.0.1.21>;tag=as25711cfe
Call-ID: a50a6fd9-a559a6ce@10.0.1.41
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="5e7017fd",uri="sip:3102@10.0.1.21",algorithm=MD5,response="c6b9190d5973410f96f33b5ef23f87df"
Contact: <sip:3101@10.0.1.41:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog 'a50a6fd9-a559a6ce@10.0.1.41' Method: ACK
akira*CLI>