<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '45827a0f4a0bd3fd2a41615126d0daff@206.123.123.123' Method: OPTIONS
asteriskserver*CLI> 
<--- SIP read from UDP://111.222.333.444:5060 --->
INVITE sip:2141234567@206.123.123.123 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Supported: timer, 100rel
To: <sip:2141234567@206.123.123.123>
From: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
P-Asserted-Identity:<sip:8009801234@10.103.102.22>
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bKce630dd00f1a57db1f9035c827c4b151
Contact: <sip:8009801234@111.222.333.444:5060>
Expires: 330
Call-Info: <sip:111.222.333.444>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 265

v=0
o=gsbc04-lsan 315845633 1261427155 IN IP4 111.222.333.444
s=sip call
c=IN IP4 66.42.119.4
t=0 0
m=audio 26224 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
--- (16 headers 12 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Sending to 111.222.333.444 : 5060 (no NAT)
Using INVITE request as basis request - 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
Found peer 'sipprovider' for '8009801234' from 111.222.333.444:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.42.119.4:26224
Looking for 2141234567 in incoming (domain 206.123.123.123)
list_route: hop: <sip:8009801234@111.222.333.444:5060>
asteriskserver*CLI> 
<--- Transmitting (no NAT) to 111.222.333.444:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bKce630dd00f1a57db1f9035c827c4b151;received=111.222.333.444
From: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
To: <sip:2141234567@206.123.123.123>
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.12
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uac
Contact: <sip:2141234567@206.123.123.123>
Content-Length: 0


<------------>
    -- Executing [2141234567@incoming:1] Goto("SIP/sipprovider-00000002", "phones,116,1") in new stack
    -- Goto (phones,116,1)
    -- Executing [116@phones:1] Answer("SIP/sipprovider-00000002", "") in new stack
Audio is at 206.123.123.123 port 13048
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
asteriskserver*CLI> 
<--- Reliably Transmitting (no NAT) to 111.222.333.444:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bKce630dd00f1a57db1f9035c827c4b151;received=111.222.333.444
From: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
To: <sip:2141234567@206.123.123.123>;tag=as45c9f854
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.12
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uac
Contact: <sip:2141234567@206.123.123.123>
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 280535478 280535478 IN IP4 206.123.123.123
s=Asterisk PBX 1.6.1.12
c=IN IP4 206.123.123.123
t=0 0
m=audio 13048 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
asteriskserver*CLI> 
<--- SIP read from UDP://111.222.333.444:5060 --->
ACK sip:2141234567@206.123.123.123 SIP/2.0
Max-Forwards: 69
To: <sip:2141234567@206.123.123.123>;tag=as45c9f854
From: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 1 ACK
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK1ada5096bf96a294ffd8d62185a52387
Contact: <sip:8009801234@111.222.333.444:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- Executing [116@phones:2] Wait("SIP/sipprovider-00000002", "2") in new stack
    -- Executing [116@phones:3] Set("SIP/sipprovider-00000002", "CALLERID(name)=") in new stack
    -- Executing [116@phones:4] NoOp("SIP/sipprovider-00000002", "Caller ID Number = 8009801234 Caller ID Name = ") in new stack
    -- Executing [116@phones:5] GotoIf("SIP/sipprovider-00000002", "0}") in new stack
    -- Executing [116@phones:6] Set("SIP/sipprovider-00000002", "CALLERID(name)=") in new stack
    -- Executing [116@phones:7] GotoIf("SIP/sipprovider-00000002", "0?Unknown") in new stack
    -- Executing [116@phones:8] GotoIf("SIP/sipprovider-00000002", "0?Start") in new stack
    -- Executing [116@phones:9] Set("SIP/sipprovider-00000002", "CALLERID(name)=") in new stack
    -- Executing [116@phones:10] GotoIf("SIP/sipprovider-00000002", "1?Unknown:Start") in new stack
    -- Goto (phones,116,11)
    -- Executing [116@phones:11] Set("SIP/sipprovider-00000002", "CALLERID(name)=Unknown") in new stack
    -- Executing [116@phones:12] Dial("SIP/sipprovider-00000002", "SIP/116,30") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
    -- Called 116
    -- SIP/116-00000003 is ringing
    -- SIP/116-00000003 answered SIP/sipprovider-00000002
set_destination: Parsing <sip:8009801234@111.222.333.444:5060> for address/port to send to
set_destination: set destination to 111.222.333.444, port 5060
Reliably Transmitting (no NAT) to 111.222.333.444:5060:
INVITE sip:8009801234@111.222.333.444:5060 SIP/2.0
Via: SIP/2.0/UDP 206.123.123.123:5060;branch=z9hG4bK7c7245a5;rport
Max-Forwards: 70
From: <sip:2141234567@206.123.123.123>;tag=as45c9f854
To: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
Contact: <sip:2141234567@206.123.123.123>
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.12
Require: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 280535478 280535479 IN IP4 206.123.123.123
s=Asterisk PBX 1.6.1.12
c=IN IP4 206.123.123.123
t=0 0
m=image 4786 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:259
a=T38FaxUdpEC:t38UDPRedundancy

---
asteriskserver*CLI> 
<--- SIP read from UDP://111.222.333.444:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 206.123.123.123:5060;branch=z9hG4bK7c7245a5;rport
From: <sip:2141234567@206.123.123.123>;tag=as45c9f854
To: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
asteriskserver*CLI> 
<--- SIP read from UDP://111.222.333.444:5060 --->
SIP/2.0 200 OK
Session-Expires: 1800;refresher=uac
Require: timer
Via: SIP/2.0/UDP 206.123.123.123:5060;branch=z9hG4bK7c7245a5;rport
To: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
From: <sip:2141234567@206.123.123.123>;tag=as45c9f854
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 102 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:8009801234@111.222.333.444:5060>
Call-Info: <sip:111.222.333.444>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 248

v=0
o=gsbc04-lsan 315845633 1261427156 IN IP4 111.222.333.444
s=sip call
c=IN IP4 66.42.119.4
t=0 0
m=image 26224 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (13 headers 11 lines) ---
Got T.38 offer in SDP in dialog 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Dec 21 14:34:50] WARNING[13242]: udptl.c:766 calculate_far_max_ifp: (no tag): Cannot calculate far_max_ifp before far_max_datagram has been set.
set_destination: Parsing <sip:8009801234@111.222.333.444:5060> for address/port to send to
set_destination: set destination to 111.222.333.444, port 5060
Transmitting (no NAT) to 111.222.333.444:5060:
ACK sip:8009801234@111.222.333.444:5060 SIP/2.0
Via: SIP/2.0/UDP 206.123.123.123:5060;branch=z9hG4bK31af9c9f;rport
Max-Forwards: 70
From: <sip:2141234567@206.123.123.123>;tag=as45c9f854
To: <sip:8009801234@111.222.333.444>;tag=3470416482-301237
Contact: <sip:2141234567@206.123.123.123>
Call-ID: 78089018-3470416482-301228@gsbc04-lsan.mdsg-sipprovider.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.12
Content-Length: 0


---
[Dec 21 14:34:50] WARNING[13288]: udptl.c:725 calculate_local_max_datagram: (SIP/116): Cannot calculate local_max_datagram before local_max_ifp has been set.
asteriskserver*CLI> 
Disconnected from Asterisk server
/usr/sbin/safe_asterisk: line 146: 13221 Segmentation fault      (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
[root@asteriskserver asterisk]# Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed