rendered paste body--- SIP read from 8.8.194.3:5065 --->
INVITE sip:7119765421@204.126.120.201:5060 SIP/2.0
From: "Asterisk TestPhone" <sip:5026382680@lycin.net>;tag=8af60d5c-1dd2-11b2-bcc4-b03162323164+8afe005c
Contact: "Asterisk TestPhone" <sip:5026382680@8.8.194.3:5065;transport=udp>
To: <sip:7119765421@204.126.120.201>
Call-ID: 59998-1944843129@8.8.194.3
CSeq: 1 INVITE
Via: SIP/2.0/UDP 8.8.194.3:5065;branch=z9hG4bK1913608549512361
Max-Forwards: 70
Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY
Content-Type: application/sdp
Content-Length: 195
v=0
o=- 3381628222 3381628222 IN IP4 8.8.194.3
s=MGCP Call
c=IN IP4 65.89.123.10
t=0 0
m=audio 16704 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (11 headers 9 lines) ---
Sending to 8.8.194.3 : 5065 (no NAT)
Using INVITE request as basis request - 59998-1944843129@8.8.194.3
[Dec 11 09:15:36] DEBUG[6132]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine.
[Dec 11 09:15:36] DEBUG[6132]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '5026382680'
Found user '5026382680'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 65.89.123.10:16704
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 65.89.123.10:16704
Looking for 7119765421 in lyvmail-test (domain 204.126.120.201)
list_route: hop: <sip:5026382680@8.8.194.3:5065;transport=udp>
<--- Transmitting (no NAT) to 8.8.194.3:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 8.8.194.3:5065;branch=z9hG4bK1913608549512361;received=8.8.194.3
From: "Asterisk TestPhone" <sip:5026382680@lycin.net>;tag=8af60d5c-1dd2-11b2-bcc4-b03162323164+8afe005c
To: <sip:7119765421@204.126.120.201>
Call-ID: 59998-1944843129@8.8.194.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7119765421@204.126.120.201>
Content-Length: 0
<------------>
[Dec 11 09:15:36] DEBUG[6110]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine.
[Dec 11 09:15:36] DEBUG[6110]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '5026382680'
-- Executing [7119765421@lyvmail-test:1] Goto("SIP/5026382680-007f5840", "lyvmail-test-run|s|1") in new stack
-- Goto (lyvmail-test-run,s,1)
-- Executing [s@lyvmail-test-run:1] Answer("SIP/5026382680-007f5840", "") in new stack
Audio is at 204.126.120.201 port 14302
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI>
<--- Reliably Transmitting (no NAT) to 8.8.194.3:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 8.8.194.3:5065;branch=z9hG4bK1913608549512361;received=8.8.194.3
From: "Asterisk TestPhone" <sip:5026382680@lycin.net>;tag=8af60d5c-1dd2-11b2-bcc4-b03162323164+8afe005c
To: <sip:7119765421@204.126.120.201>;tag=as3f0a9b38
Call-ID: 59998-1944843129@8.8.194.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7119765421@204.126.120.201>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 6030 6030 IN IP4 204.126.120.201
s=session
c=IN IP4 204.126.120.201
t=0 0
m=audio 14302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Executing [s@lyvmail-test-run:2] BackGround("SIP/5026382680-007f5840", "/var/lib/asterisk/sounds/mysounds/sounds/go-away2") in new stack
-- <SIP/5026382680-007f5840> Playing '/var/lib/asterisk/sounds/mysounds/sounds/go-away2' (language 'en')
localhost*CLI>