rendered paste body--- (12 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 192.168.0.87:5060:
INVITE sip:401@192.168.0.95 SIP/2.0
To: <sip:401@192.168.0.95>
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-fc57b040ef2eb31f-1--d87543-;rport
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 1 INVITE
Contact: <sip:402@192.168.0.87:5060>
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: Softphone2
Content-Length: 272
v=0
o=- 0 0 IN IP4 192.168.0.87
s=
c=IN IP4 192.168.0.87
t=0 0
m=audio 5000 RTP/AVP 3 97 110 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--- (11 headers 13 lines)---
Using INVITE request as basis request - 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
Sending to 192.168.0.87 : 5060 (NAT)
Reliably Transmitting (no NAT) to 192.168.0.87:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-fc57b040ef2eb31f-1--d87543-;rport;received=192.168.0.87
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
To: <sip:401@192.168.0.95>;tag=as4607b9c8
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:401@192.168.0.95>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b51d48c"
Content-Length: 0
---
Scheduling destruction of call '8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs' in 15000 ms
Found user '402'
asterisk1*CLI>
<-- SIP read from 192.168.0.87:5060:
ACK sip:401@192.168.0.95 SIP/2.0
To: <sip:401@192.168.0.95>;tag=as4607b9c8
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-fc57b040ef2eb31f-1--d87543-;rport
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 192.168.0.87:5060:
INVITE sip:401@192.168.0.95 SIP/2.0
To: <sip:401@192.168.0.95>
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 2 INVITE
Contact: <sip:402@192.168.0.87:5060>
Max-Forwards: 70
Content-Type: application/sdp
Proxy-Authorization: Digest username="402",realm="asterisk",nonce="4b51d48c",uri="sip:401@192.168.0.95",response="06d693277c83b9599fba9001db9e339a",algorithm=MD5
User-Agent: Softphone2
Content-Length: 272
v=0
o=- 0 0 IN IP4 192.168.0.87
s=
c=IN IP4 192.168.0.87
t=0 0
m=audio 5000 RTP/AVP 3 97 110 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--- (12 headers 13 lines)---
Using INVITE request as basis request - 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
Sending to 192.168.0.87 : 5060 (NAT)
Found user '402'
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.87:5000
Found description format GSM
Found description format iLBC
Found description format SPEEX
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 401 in from-internal (domain 192.168.0.95)
list_route: hop: <sip:402@192.168.0.87:5060>
Transmitting (no NAT) to 192.168.0.87:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport;received=192.168.0.87
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
To: <sip:401@192.168.0.95>
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:401@192.168.0.95>
Content-Length: 0
---
recordingcheck|20061212-112332|1165940612.50: Inbound recording not enabled
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Eric Martindale (SIP)' number is '402'
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Extension 401 is not available to be called
dialparties.agi: Extension 401 has call waiting disabled
Transmitting (no NAT) to 192.168.0.87:5060:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport;received=192.168.0.87
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
To: <sip:401@192.168.0.95>;tag=as119757a6
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:401@192.168.0.95>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.0.87:5060:
ACK sip:401@192.168.0.95 SIP/2.0
To: <sip:401@192.168.0.95>;tag=as119757a6
From: "402"<sip:402@192.168.0.95>;tag=8d10bb3d
Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport
Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs
CSeq: 2 ACK
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call '8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs'