rendered paste bodylocalhost*CLI>
<--- SIP read from 192.168.1.21:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
localhost*CLI>
<--- SIP read from 192.168.1.8:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
localhost*CLI>
<--- SIP read from 192.168.1.21:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
localhost*CLI>
<--- SIP read from 192.168.1.8:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'uccgserv' logged on from 192.168.1.20
Audio is at 192.168.1.55 port 15964
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.21:5060 :
INVITE sip:1000@192.168.1.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport
From: "1000" < sip:1000@192.168.1.55>;tag=as7cdeb343
To: <sip:1000@192.168.1.21>
Contact: <sip:1000@192.168.1.55 >
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 04 Apr 2007 18:00:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 26722 26722 IN IP4 192.168.1.55
s=session
c=IN IP4 192.168.1.55
t=0 0
m=audio 15964 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
localhost*CLI>
<--- SIP read from 192.168.1.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport
From: "1000" <sip:1000@192.168.1.55>;tag=as7cdeb343
To: <sip:1000@192.168.1.21 >
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 102 INVITE
User-Agent: Grandstream BT120 1.0.8.17
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 192.168.1.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport
From: "1000" <sip:1000@192.168.1.55>;tag=as7cdeb343
To: < sip:1000@192.168.1.21>;tag=b8682c6199346e89
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 102 INVITE
User-Agent: Grandstream BT120 1.0.8.17
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'uccgserv' logged on from 192.168.1.20
== Manager 'uccgserv' logged off from 192.168.1.20
localhost*CLI>
<--- SIP read from 192.168.1.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport
From: "1000" < sip:1000@192.168.1.55>;tag=as7cdeb343
To: <sip:1000@192.168.1.21>;tag=b8682c6199346e89
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 102 INVITE
User-Agent: Grandstream BT120 1.0.8.17
Contact: <sip:1000@192.168.1.21 >
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 155
v=0
o=1000 8000 8000 IN IP4 192.168.1.21
s=SIP Call
c=IN IP4 192.168.1.21
t=0 0
m=audio 5004 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 192.168.1.21:5004
Found description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.21:5004
list_route: hop: < sip:1000@192.168.1.21>
set_destination: Parsing <sip:1000@192.168.1.21> for address/port to send to
set_destination: set destination to 192.168.1.21, port 5060
Transmitting (no NAT) to 192.168.1.21:5060:
ACK sip:1000@192.168.1.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK4a3f2985;rport
From: "1000" <sip:1000@192.168.1.55>;tag=as7cdeb343
To: <sip:1000@192.168.1.21 >;tag=b8682c6199346e89
Contact: <sip:1000@192.168.1.55>
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
> Channel SIP/1000-087a05b8 was answered.
== Manager 'uccgserv' logged off from 192.168.1.20
== Starting SIP/1000-087a05b8 at outbound,2000,1 failed so falling back to exten 's'
== Starting SIP/1000-087a05b8 at outbound,s,1 still failed so falling back to context 'default'
[Apr 4 14:00:39] WARNING[27497]: pbx.c:2405 __ast_pbx_run: Channel 'SIP/1000-087a05b8' sent into invalid extension 's' in context 'default', but no invalid handler
Scheduling destruction of SIP dialog ' 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:1000@192.168.1.21 > for address/port to send to
set_destination: set destination to 192.168.1.21, port 5060
Reliably Transmitting (no NAT) to 192.168.1.21:5060 :
BYE sip:1000@192.168.1.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK237c155b;rport
rom: "1000" < sip:1000@192.168.1.55>;tag=as7cdeb343
To: <sip:1000@192.168.1.21>;tag=b8682c6199346e89
Contact: < sip:1000@192.168.1.55>
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
localhost*CLI>
<--- SIP read from 192.168.1.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060 ;branch=z9hG4bK237c155b;rport
From: "1000" <sip:1000@192.168.1.55>;tag=as7cdeb343
To: <sip:1000@192.168.1.21 >;tag=b8682c6199346e89
Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55
CSeq: 103 BYE
User-Agent: Grandstream BT120 1.0.8.17
Contact: <sip:1000@192.168.1.21>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55' Method: INVITE
localhost*CLI>
<--- SIP read from 192.168.1.21:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
localhost*CLI>
<--- SIP read from 192.168.1.8:5060 --->