All pastes #732053 Raw Edit

Unnamed

public text v1 · immutable
·published 2007-10-10 16:48 UTC
rendered paste body
asteriskpbx*CLI> 
<--- SIP read from 192.168.69.51:5060 --->
INVITE sip:4814900@192.168.69.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bK749b7fb7D72F3196
From: "Shawn Byrne" <sip:201@192.168.69.1>;tag=81117709-3AC186A2
To: <sip:4814900@192.168.69.1;user=phone>
CSeq: 1 INVITE
Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
Contact: <sip:201@192.168.69.51>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 978314410 978314410 IN IP4 192.168.69.51
s=Polycom IP Phone
c=IN IP4 192.168.69.51
t=0 0
m=audio 2236 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (14 headers 11 lines) ---

Sending to 192.168.69.51 : 5060 (no NAT)
Using INVITE request as basis request - bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
Found peer '201'

<--- Reliably Transmitting (NAT) to 192.168.69.51:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bK749b7fb7D72F3196;received=192.168.69.51
From: "Shawn Byrne" <sip:201@192.168.69.1>;tag=81117709-3AC186A2
To: <sip:4814900@192.168.69.1;user=phone>;tag=as721870ea
Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL: 
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d40a7f2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51' in 32000 ms (Method: INVITE)

asteriskpbx*CLI> 
<--- SIP read from 192.168.69.51:5060 --->
ACK sip:4814900@192.168.69.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bK749b7fb7D72F3196
From: "Shawn Byrne" <sip:201@192.168.69.1>;tag=81117709-3AC186A2
To: <sip:4814900@192.168.69.1;user=phone>;tag=as721870ea
CSeq: 1 ACK
Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
Contact: <sip:201@192.168.69.51>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

asteriskpbx*CLI> 
<--- SIP read from 192.168.69.51:5060 --->
INVITE sip:4814900@192.168.69.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bKfe8f23985460AC0F
From: "Shawn Byrne" <sip:201@192.168.69.1>;tag=81117709-3AC186A2
To: <sip:4814900@192.168.69.1;user=phone>
CSeq: 2 INVITE
Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
Contact: <sip:201@192.168.69.51>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="201", realm="asterisk", nonce="3d40a7f2", uri="sip:4814900@192.168.69.1:5060;user=phone", response="208e082dbd129c41e9e27c14f73e2e19", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 978314410 978314410 IN IP4 192.168.69.51
s=Polycom IP Phone
c=IN IP4 192.168.69.51
t=0 0
m=audio 2236 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a
=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.69.51 : 5060 (NAT)
Using INVITE request as basis request - bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
Found peer '201'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.69.51:2236
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.69.51:2236
Looking for 4814900 in numberplan-custom-1 (domain 192.168.69.1)

<--- Reliably Transmitting (NAT) to 192.168.69.51:5060 --->

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bKfe8f23985460AC0F;received=192.168.69.51
From: "Shawn Byrne" <sip:201@192.168.69.1>;tag=81117709-3AC186A2
To: <sip:4814900@192.168.69.1;user=phone>;tag=as721870ea
Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL: 
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51' in 32000 ms (Method: INVITE)

asteriskpbx*CLI> 
<--- SIP read from 192.168.69.51:5060 --->
ACK sip:4814900@192.168.69.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bKfe8f23985460AC0F
From: "Shawn Byrne" <sip:201@192.168.69.1>;tag=81117709-3AC186A2
To: <sip:4814900@192.168.69.1;user=phone>;tag=as721870ea
CSeq: 2 ACK
Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51
Contact: <sip:201@192.168.69.51>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078
Proxy-Authorization: Digest username="201", realm="asterisk", nonce="3d40a7f2", uri="sip:4814900@192.168.69.1:5060;user=phone", response="208e082dbd129c41e9e27c14f73e2e19", algorithm=MD5
Max-Forwards: 70
Content-Length: 0