Call is disconnected after exactly 10 seconds

doncorley
Joined: Tue 21 of Sep, 2010

Call is disconnected after exactly 10 seconds

Posted:Wed 17 of Nov, 2010 (20:55 UTC)
I have a remote asterisk system that I am accessing from home. It used to work fine until I upgraded my DSL modem to a Verizon labeled ActionTEC GT704WG. All clients (X-Lite softphone and Polycom IP Phone) disconnect after exactly 10 seconds, but both clients work fine behind other DSL modems.
Any help you can give me is appreciated.
Thanks,
Don

Here is my SIP config:
[201]
deny=0.0.0.0/0.0.0.0
secret=xxxxxxxx
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/201
mailbox=201@device
permit=0.0.0.0/0.0.0.0
callerid=device <201>
callcounter=yes
faxdetect=no


Here is my asterisk debug log:

Start of call
--------------------------------------------------------------------------------------
*CLI> 
<--- SIP read from UDP:96.251.177.249:1024 --->
INVITE sip:16263582903@sip.tourgeek.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK1df29571D5378542
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>
CSeq: 1 INVITE
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
Contact: <sip:201@96.251.177.249>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 262

v=0
o=- 1289959394 1289959394 IN IP4 96.251.177.249
s=Polycom IP Phone
c=IN IP4 96.251.177.249
t=0 0
m=audio 2242 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

<------------->
--- (15 headers 11 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 96.251.177.249 : 5060 (no NAT)
Using INVITE request as basis request - 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
Found peer '201' for '201' from 96.251.177.249:1024

<--- Reliably Transmitting (NAT) to 96.251.177.249:1024 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK1df29571D5378542;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as11718f2c
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78449c46"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8f5b8c3f-ff52aebd-351494f8@192.168.1.64' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:96.251.177.249:1024 --->
ACK sip:16263582903@sip.tourgeek.com:5060 SIP/2.0
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK1df29571D5378542
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as11718f2c
CSeq: 1 ACK
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
Contact: <sip:201@96.251.177.249>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017
Accept-Language: en
Max-Forwards: 70
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:96.251.177.249:1024 --->
INVITE sip:16263582903@sip.tourgeek.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>
CSeq: 2 INVITE
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
Contact: <sip:201@96.251.177.249>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Authorization: Digest username="201", realm="asterisk", nonce="78449c46", uri="sip:16263582903@sip.tourgeek.com:5060;user=phone", response="facbbf094c0d11bcf51b0e2ce5ad20f5", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 262

v=0
o=- 1289959394 1289959394 IN IP4 96.251.177.249
s=Polycom IP Phone
c=IN IP4 96.251.177.249
t=0 0
m=audio 2242 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

<------------->
--- (16 headers 11 lines) ---
Sending to 96.251.177.249 : 1024 (NAT)
Using INVITE request as basis request - 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
Found peer '201' for '201' from 96.251.177.249:1024
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 96.251.177.249:2242
Looking for 16263582903 in from-internal (domain sip.tourgeek.com)
list_route: hop: <sip:201@96.251.177.249>

<--- Transmitting (NAT) to 96.251.177.249:1024 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Length: 0


<------------>
    -- Executing [16263582903@from-internal:1] Macro("SIP/201-00000014", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/201-00000014", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-00000014", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-00000014", "1?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/201-00000014", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/201-00000014", "AMPUSERCIDNAME=Don Corley") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-00000014", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/201-00000014", "AMPUSERCID=201") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/201-00000014", "CALLERID(all)="Don Corley" <201>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/201-00000014", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/201-00000014", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/201-00000014", "Using CallerID "Don Corley" <201>") in new stack
    -- Executing [16263582903@from-internal:2] NoOp("SIP/201-00000014", "Calling Out Route: main-route-out") in new stack
    -- Executing [16263582903@from-internal:3] Set("SIP/201-00000014", "MOH in new stack
    -- Executing [16263582903@from-internal:4] Set("SIP/201-00000014", "_NODEST=") in new stack
    -- Executing [16263582903@from-internal:5] Macro("SIP/201-00000014", "record-enable,201,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/201-00000014", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/201-00000014", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/201-00000014", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/201-00000014", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/201-00000014", "1?MacroExit()") in new stack
    -- Executing [16263582903@from-internal:6] Macro("SIP/201-00000014", "dialout-trunk,2,16263582903,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/201-00000014", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/201-00000014", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/201-00000014", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/201-00000014", "DIAL_NUMBER=16263582903") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/201-00000014", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/201-00000014", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/201-00000014", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/201-00000014", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/201-00000014", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/201-00000014", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/201-00000014", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/201-00000014", "0?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/201-00000014", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/201-00000014", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/201-00000014", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/201-00000014", "TRUNKOUTCID="tourgeek.com" <16262057017>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/201-00000014", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/201-00000014", "1?Set(CALLERID(all)=tourgeek.com <16262057017>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/201-00000014", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/201-00000014", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/201-00000014", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/201-00000014", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/201-00000014", "OUTNUM=16263582903") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/201-00000014", "custom=SIP/vitel-outbound") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/201-00000014", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/201-00000014", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/201-00000014", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/201-00000014", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/201-00000014", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/201-00000014", "SIP/vitel-outbound/16263582903,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
We think we can do text
Audio is at 67.227.17.86 port 10898
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.2.142.93:5060:
INVITE sip:16263582903@64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP 67.227.17.86:5060;branch=z9hG4bK3efa4306;rport
Max-Forwards: 70
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>
Contact: <sip:16262057017@67.227.17.86>
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.14
Date: Thu, 18 Nov 2010 03:56:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 445

v=0
o=root 229857003 229857003 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 10898 RTP/AVP 0 8 3 112 5 10 7 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called vitel-outbound/16263582903

<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 67.227.17.86:5060;branch=z9hG4bK3efa4306;rport=5060
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 67.227.17.86:5060;received=67.227.17.86;branch=z9hG4bK3efa4306;rport=5060
Record-Route: <sip:64.2.142.93;lr=on;did=c3e.a0e28371>
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>;tag=as37968994
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:16263582903@64.2.142.195:5060>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15507 15507 IN IP4 64.2.142.195
s=session
c=IN IP4 64.2.142.195
t=0 0
m=audio 18586 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.2.142.195:18586
    -- SIP/vitel-outbound-00000015 is making progress passing it to SIP/201-00000014
Audio is at 67.227.17.86 port 14414
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 96.251.177.249:1024 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050931 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
--------------------------------------------------------------------------------------
Phone is ringing
--------------------------------------------------------------------------------------
<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.227.17.86:5060;received=67.227.17.86;branch=z9hG4bK3efa4306;rport=5060
Record-Route: <sip:64.2.142.93;lr=on;did=c3e.a0e28371>
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>;tag=as37968994
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:16263582903@64.2.142.195:5060>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 15507 15508 IN IP4 64.2.142.195
s=session
c=IN IP4 64.2.142.195
t=0 0
m=audio 18586 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.2.142.195:18586
list_route: hop: <sip:64.2.142.93;lr=on;did=c3e.a0e28371>
set_destination: Parsing <sip:64.2.142.93;lr=on;did=c3e.a0e28371> for address/port to send to
set_destination: set destination to 64.2.142.93, port 5060
Transmitting (no NAT) to 64.2.142.93:5060:
ACK sip:16263582903@64.2.142.195:5060 SIP/2.0
Via: SIP/2.0/UDP 67.227.17.86:5060;branch=z9hG4bK2ff10033;rport
Route: <sip:64.2.142.93;lr=on;did=c3e.a0e28371>
Max-Forwards: 70
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>;tag=as37968994
Contact: <sip:16262057017@67.227.17.86>
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.14
Content-Length: 0


---
    -- SIP/vitel-outbound-00000015 answered SIP/201-00000014
Audio is at 67.227.17.86 port 14414
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 96.251.177.249:1024 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
--------------------------------------------------------------------------------------
Phone answers
--------------------------------------------------------------------------------------

<------------>
    -- Packet2Packet bridging SIP/201-00000014 and SIP/vitel-outbound-00000015
Retransmitting #1 (NAT) to 96.251.177.249:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 96.251.177.249:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (NAT) to 96.251.177.249:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #4 (NAT) to 96.251.177.249:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #5 (NAT) to 96.251.177.249:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:96.251.177.249:17818 --->



<------------->
--------------------------------------------------------------------------------------
Talking on Phone
--------------------------------------------------------------------------------------

Retransmitting #6 (NAT) to 96.251.177.249:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.251.177.249;branch=z9hG4bK7a737bb41A5540C9;received=96.251.177.249
From: "Don" <sip:201@sip.tourgeek.com>;tag=807FFC6B-8FCFCE9E
To: <sip:16263582903@sip.tourgeek.com;user=phone>;tag=as3e7f6b8f
Call-ID: 8f5b8c3f-ff52aebd-351494f8@192.168.1.64
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:16263582903@67.227.17.86>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 291050931 291050932 IN IP4 67.227.17.86
s=Asterisk PBX 1.6.2.14
c=IN IP4 67.227.17.86
t=0 0
m=audio 14414 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/201-00000014", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-00000014", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-00000014", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-00000014", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-00000014", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-00000014' in macro 'hangupcall'
Scheduling destruction of SIP dialog '7085bdf325824d893ea5664570f53da3@67.227.17.86' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:64.2.142.93;lr=on;did=c3e.a0e28371> for address/port to send to
set_destination: set destination to 64.2.142.93, port 5060
Reliably Transmitting (no NAT) to 64.2.142.93:5060:
BYE sip:16263582903@64.2.142.195:5060 SIP/2.0
Via: SIP/2.0/UDP 67.227.17.86:5060;branch=z9hG4bK3058eecb;rport
Route: <sip:64.2.142.93;lr=on;did=c3e.a0e28371>
Max-Forwards: 70
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>;tag=as37968994
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.14
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/201-00000014' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 16263582903, 6) exited non-zero on 'SIP/201-00000014'

<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.227.17.86:5060;received=67.227.17.86;branch=z9hG4bK3058eecb;rport=5060
From: "tourgeek.com" <sip:16262057017@67.227.17.86>;tag=as7810fe2a
To: <sip:16263582903@64.2.142.93>;tag=as37968994
Call-ID: 7085bdf325824d893ea5664570f53da3@67.227.17.86
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7085bdf325824d893ea5664570f53da3@67.227.17.86' Method: INVITE
Really destroying SIP dialog '8f5b8c3f-ff52aebd-351494f8@192.168.1.64' Method: INVITE
--------------------------------------------------------------------------------------
Voice stops
--------------------------------------------------------------------------------------

Reliably Transmitting (NAT) to 96.251.177.249:17818: