rendered paste bodyWITH CALLER ID DISABLED
pbx1*CLI>
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
INVITE sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK1084469646
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>
Call-ID: 1112494159@10.10.10.39
CSeq: 1 INVITE
Contact: <sip:3000@10.10.10.39>
Max-Forwards: 70
User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.)
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER
Content-Type: application/sdp
Content-Length: 195
v=0
o=3000 123456 654321 IN IP4 10.10.10.39
s=A conversation
c=IN IP4 10.10.10.39
t=0 0
m=audio 12080 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
--- (12 headers 9 lines)---
Using INVITE request as basis request - 1112494159@10.10.10.39
Sending to 10.10.10.39 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.10.39:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK1084469646;received=10.10.10.39
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>;tag=as6f5cbdc6
Call-ID: 1112494159@10.10.10.39
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.10.10>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3eb3c174"
Content-Length: 0
---
Scheduling destruction of call '1112494159@10.10.10.39' in 15000 ms
Found user '3000'
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
ACK sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK1084469646
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>;tag=as6f5cbdc6
Call-ID: 1112494159@10.10.10.39
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines)---
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
INVITE sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3334702713
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>
Call-ID: 1112494159@10.10.10.39
CSeq: 2 INVITE
Contact: <sip:3000@10.10.10.39>
Proxy-Authorization: Digest username="3000", realm="asterisk", nonce="3eb3c174", uri="sip:s@10.10.10.10", response="43f1c0cfdfe2bd6457f93696da56a557", algorithm=MD5
Max-Forwards: 70
User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.)
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER
Content-Type: application/sdp
Content-Length: 195
v=0
o=3000 123456 654321 IN IP4 10.10.10.39
s=A conversation
c=IN IP4 10.10.10.39
t=0 0
m=audio 12080 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
--- (13 headers 9 lines)---
Using INVITE request as basis request - 1112494159@10.10.10.39
Sending to 10.10.10.39 : 5060 (non-NAT)
Found user '3000'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.10.10.39:12080
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in inbound (domain 10.10.10.10)
list_route: hop: <sip:3000@10.10.10.39>
Transmitting (no NAT) to 10.10.10.39:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3334702713;received=10.10.10.39
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>
Call-ID: 1112494159@10.10.10.39
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.10.10>
Content-Length: 0
---
-- Executing Wait("SIP/3000-0815a048", "2") in new stack
-- Executing Answer("SIP/3000-0815a048", "") in new stack
We're at 10.10.10.10 port 12508
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.10.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3334702713;received=10.10.10.39
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>;tag=as06e5c823
Call-ID: 1112494159@10.10.10.39
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.10.10>
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 32372 32372 IN IP4 10.10.10.10
s=session
c=IN IP4 10.10.10.10
t=0 0
m=audio 12508 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing NoOp("SIP/3000-0815a048", ""3000" <3000> 3000 3000") in new stack
-- Executing BackGround("SIP/3000-0815a048", "726_welcome") in new stack
-- Playing '726_welcome' (language 'en')
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
ACK sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK398548864
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>;tag=as06e5c823
Call-ID: 1112494159@10.10.10.39
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.)
Content-Length: 0
--- (9 headers 0 lines)---
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
BYE sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3381027283
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>;tag=as06e5c823
Call-ID: 1112494159@10.10.10.39
CSeq: 3 BYE
Max-Forwards: 70
User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.)
Content-Length: 0
--- (9 headers 0 lines)---
Sending to 10.10.10.39 : 5060 (non-NAT)
Transmitting (no NAT) to 10.10.10.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3381027283;received=10.10.10.39
From: 3000<sip:3000@10.10.10.10>;tag=3815248908
To: <sip:s@10.10.10.10>;tag=as06e5c823
Call-ID: 1112494159@10.10.10.39
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.10.10>
Content-Length: 0
---
== Spawn extension (inbound, s, 4) exited non-zero on 'SIP/3000-0815a048'
-- Executing Hangup("SIP/3000-0815a048", "") in new stack
== Spawn extension (inbound, h, 1) exited non-zero on 'SIP/3000-0815a048'
Destroying call '1112494159@10.10.10.39'
pbx1*CLI>
=========================================================================================
=========================================================================================
=========================================================================================
=========================================================================================
=========================================================================================
WITH CALLER ID ENABLED
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
INVITE sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK654133672
From: 3000<sip:3523029577@10.10.10.10>;tag=249097462
To: <sip:s@10.10.10.10>
Call-ID: 1390462977@10.10.10.39
CSeq: 1 INVITE
Contact: <sip:3523029577@10.10.10.39>
Max-Forwards: 70
User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.)
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER
Content-Type: application/sdp
Content-Length: 195
v=0
o=3000 123456 654321 IN IP4 10.10.10.39
s=A conversation
c=IN IP4 10.10.10.39
t=0 0
m=audio 12080 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
--- (12 headers 9 lines)---
Using INVITE request as basis request - 1390462977@10.10.10.39
Sending to 10.10.10.39 : 5060 (non-NAT)
Found peer '3030'
Reliably Transmitting (no NAT) to 10.10.10.39:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK654133672;received=10.10.10.39
From: 3000<sip:3523029577@10.10.10.10>;tag=249097462
To: <sip:s@10.10.10.10>;tag=as0b3678cf
Call-ID: 1390462977@10.10.10.39
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.10.10>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1aecd999"
Content-Length: 0
---
Scheduling destruction of call '1390462977@10.10.10.39' in 15000 ms
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
ACK sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK654133672
From: 3000<sip:3523029577@10.10.10.10>;tag=249097462
To: <sip:s@10.10.10.10>;tag=as0b3678cf
Call-ID: 1390462977@10.10.10.39
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines)---
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
INVITE sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK2531879195
From: 3000<sip:3523029577@10.10.10.10>;tag=249097462
To: <sip:s@10.10.10.10>
Call-ID: 1390462977@10.10.10.39
CSeq: 2 INVITE
Contact: <sip:3523029577@10.10.10.39>
Proxy-Authorization: Digest username="3000", realm="asterisk", nonce="1aecd999", uri="sip:s@10.10.10.10", response="2c121175fb4b01fd34373e934ade9792", algorithm=MD5
Max-Forwards: 70
User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.)
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER
Content-Type: application/sdp
Content-Length: 195
v=0
o=3000 123456 654321 IN IP4 10.10.10.39
s=A conversation
c=IN IP4 10.10.10.39
t=0 0
m=audio 12080 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000/1
--- (13 headers 9 lines)---
Using INVITE request as basis request - 1390462977@10.10.10.39
Sending to 10.10.10.39 : 5060 (non-NAT)
Found peer '3030'
Nov 3 14:03:00 NOTICE[3415]: chan_sip.c:10468 handle_request_invite: Failed to authenticate user 3000<sip:3523029577@10.10.10.10>;tag=249097462
Reliably Transmitting (no NAT) to 10.10.10.39:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK2531879195;received=10.10.10.39
From: 3000<sip:3523029577@10.10.10.10>;tag=249097462
To: <sip:s@10.10.10.10>;tag=as0b3678cf
Call-ID: 1390462977@10.10.10.39
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.10.10>
Content-Length: 0
---
pbx1*CLI>
<-- SIP read from 10.10.10.39:5060:
ACK sip:s@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK2531879195
From: 3000<sip:3523029577@10.10.10.10>;tag=249097462
To: <sip:s@10.10.10.10>;tag=as0b3678cf
Call-ID: 1390462977@10.10.10.39
CSeq: 2 ACK
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call '1390462977@10.10.10.39'