-- Executing [016163090708@PBX:1] Goto("SIP/ip-teles-a-1a882a00", "004316163090708|1") in new stack
    -- Goto (PBX,004316163090708,1)
    -- Executing [004316163090708@PBX:1] Dial("SIP/ip-teles-a-1a882a00", "SIP/61054316163090708@ip-teles-b|90") in new stack
Audio is at ip-asterisk port 19212
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to ip-teles-b:5060:
INVITE sip:61054316163090708@ip-teles-b SIP/2.0
Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK48fb056f;rport
From: "33" <sip:33@ip-asterisk>;tag=as64ed608f
To: <sip:61054316163090708@ip-teles-b>
Contact: <sip:33@ip-asterisk>
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 102 INVITE
User-Agent: GlobalCall+T.38
Max-Forwards: 70
Remote-Party-ID: "33" <sip:33@ip-asterisk>;privacy=off;screen=yes
Date: Thu, 26 Mar 2009 16:28:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 23076 23076 IN IP4 ip-asterisk
s=session
c=IN IP4 ip-asterisk
t=0 0
m=audio 19212 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 61054316163090708@ip-teles-b
DataNode1*CLI>
<--- SIP read from ip-teles-b:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK48fb056f;rport
From: "33" <sip:33@ip-asterisk>;tag=as64ed608f
To: <sip:61054316163090708@ip-teles-b>
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER
User-Agent: TELES.VoIPGATE 13.0p 900
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
DataNode1*CLI>
<--- SIP read from ip-teles-b:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK48fb056f;rport
From: "33" <sip:33@ip-asterisk>;tag=as64ed608f
To: <sip:61054316163090708@ip-teles-b>;tag=705984975818541799090680920393
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER
Contact: <sip:61054316163090708@ip-teles-b>
User-Agent: TELES.VoIPGATE 13.0p 900
Content-Type: application/sdp
Content-Length: 225

v=0
o=- 173 1 IN IP4 ip-teles-b
s=-
c=IN IP4 ip-teles-b
t=0 0
m=audio 29000 RTP/AVP 18 101
a=sendrecv
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port ip-teles-b:29000
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port ip-teles-b:29000
    -- SIP/ip-teles-b-1a87e780 is making progress passing it to SIP/ip-teles-a-1a882a00
DataNode1*CLI>
<--- SIP read from ip-teles-b:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK48fb056f;rport
From: "33" <sip:33@ip-asterisk>;tag=as64ed608f
To: <sip:61054316163090708@ip-teles-b>;tag=705984975818541799090680920393
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER
Contact: <sip:61054316163090708@ip-teles-b>
User-Agent: TELES.VoIPGATE 13.0p 900
Content-Type: application/sdp
Content-Length: 225

v=0
o=- 173 1 IN IP4 ip-teles-b
s=-
c=IN IP4 ip-teles-b
t=0 0
m=audio 29000 RTP/AVP 18 101
a=sendrecv
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port ip-teles-b:29000
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port ip-teles-b:29000
    -- SIP/ip-teles-b-1a87e780 is ringing
    -- SIP/ip-teles-b-1a87e780 is making progress passing it to SIP/ip-teles-a-1a882a00
DataNode1*CLI>
<--- SIP read from ip-teles-b:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK48fb056f;rport
From: "33" <sip:33@ip-asterisk>;tag=as64ed608f
To: <sip:61054316163090708@ip-teles-b>;tag=705984975818541799090680920393
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER
Contact: <sip:61054316163090708@ip-teles-b>
User-Agent: TELES.VoIPGATE 13.0p 900
Content-Type: application/sdp
Content-Length: 225

v=0
o=- 173 1 IN IP4 ip-teles-b
s=-
c=IN IP4 ip-teles-b
t=0 0
m=audio 29000 RTP/AVP 18 101
a=sendrecv
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port ip-teles-b:29000
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port ip-teles-b:29000
list_route: hop: <sip:61054316163090708@ip-teles-b>
set_destination: Parsing <sip:61054316163090708@ip-teles-b> for address/port to send to
set_destination: set destination to ip-teles-b, port 5060
Transmitting (no NAT) to ip-teles-b:5060:
ACK sip:61054316163090708@ip-teles-b SIP/2.0
Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK13c45d43;rport
From: "33" <sip:33@ip-asterisk>;tag=as64ed608f
To: <sip:61054316163090708@ip-teles-b>;tag=705984975818541799090680920393
Contact: <sip:33@ip-asterisk>
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 102 ACK
User-Agent: GlobalCall+T.38
Max-Forwards: 70
Remote-Party-ID: "33" <sip:33@ip-asterisk>;privacy=off;screen=yes
Content-Length: 0


---
    -- SIP/ip-teles-b-1a87e780 answered SIP/ip-teles-a-1a882a00
    -- Packet2Packet bridging SIP/ip-teles-a-1a882a00 and SIP/ip-teles-b-1a87e780
DataNode1*CLI>
<--- SIP read from ip-teles-b:5060 --->
BYE sip:33@ip-asterisk SIP/2.0
Max-Forwards: 50
Via: SIP/2.0/UDP ip-teles-b:5060;rport;branch=z9hG4bK51331770808760388901264
From: <sip:61054316163090708@ip-teles-b>;tag=705984975818541799090680920393
To: "33" <sip:33@ip-asterisk>;tag=as64ed608f
Contact: <sip:61054316163090708@ip-teles-b>
User-Agent: TELES.VoIPGATE 13.0p 900
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 103 BYE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to ip-teles-b : 5060 (NAT)
DataNode1*CLI>
<--- Transmitting (NAT) to ip-teles-b:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip-teles-b:5060;branch=z9hG4bK51331770808760388901264;received=ip-teles-b;rport=5060
From: <sip:61054316163090708@ip-teles-b>;tag=705984975818541799090680920393
To: "33" <sip:33@ip-asterisk>;tag=as64ed608f
Call-ID: 2b88fc766e82335b39714b9d039fd747@ip-asterisk
CSeq: 103 BYE
User-Agent: GlobalCall+T.38
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
  == Spawn extension (PBX, 004316163090708, 1) exited non-zero on 'SIP/ip-teles-a-1a882a00'
Really destroying SIP dialog '2b88fc766e82335b39714b9d039fd747@ip-asterisk' Method: BYE
[Mar 26 17:28:17] WARNING[23094]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info.
DataNode1*CLI>