set debug 10
Core debug was 0 and is now 10
The 'set debug' command is deprecated and will be removed in a future release. Please use 'core set debug' instead.
*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
*CLI>     -- Accepting AUTHENTICATED call from 69.71.222.196:
       > requested format = ulaw,
       > requested prefs = (gsm|ilbc|ulaw|alaw|speex),
       > actual format = gsm,
       > host prefs = (gsm|ilbc),
       > priority = mine
    -- Executing [222@softphone:1] Dial("IAX2/voovox-7174", "SIP/34672523701@voovox-outbound") in new stack
Audio is at 192.168.1.3 port 19196
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.4.81.180:5060:
INVITE sip:34672523701@eugw.ast.voovox.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK77b5f62b;rport
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>
Contact: <sip:463910@192.168.1.3>
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 13 Jun 2009 20:46:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 9308 9308 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 19196 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 34672523701@voovox-outbound

<--- SIP read from 62.4.81.180:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK77b5f62b;rport
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>;tag=as7b5cc2b0
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="eugw.ast.voovox.com", nonce="0db80fa1", algorithm=MD5
user-agent: SIP gate
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 62.4.81.180:5060:
ACK sip:34672523701@eugw.ast.voovox.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK77b5f62b;rport
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>;tag=as7b5cc2b0
Contact: <sip:463910@192.168.1.3>
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 192.168.1.3 port 19196
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.4.81.180:5060:
INVITE sip:34672523701@eugw.ast.voovox.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK53be5e44;rport
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>
Contact: <sip:463910@192.168.1.3>
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="463910", realm="eugw.ast.voovox.com", algorithm=MD5, uri="sip:34672523701@eugw.ast.voovox.com", nonce="0db80fa1", response="c70c605d58380ac96d26ddf2b504359c"
Date: Sat, 13 Jun 2009 20:46:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 9308 9309 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 19196 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 62.4.81.180:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK53be5e44;rport
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 103 INVITE
Contact: <sip:34672523701@62.4.81.180>
user-agent: SIP gate
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


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

<--- SIP read from 192.168.1.2:35288 --->
SUBSCRIBE sip:1000@192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:35288;branch=z9hG4bK-d8754z-02046d111ec27d57-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@192.168.1.2:35288>
To: "asterisk"<sip:1000@192.168.1.3>
From: "asterisk"<sip:1000@192.168.1.3>;tag=27dad404
Call-ID: NjY4MzBlYmYzZmFmNzcyYzVmMzk4OTIwM2Q5OWMxYWI.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1014k stamp 47051
Event: message-summary
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.2 : 35288 (NAT)
Found peer '1000'
Looking for 1000 in adhearsion (domain 192.168.1.3)

<--- Transmitting (no NAT) to 192.168.1.2:35288 --->
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.2:35288;branch=z9hG4bK-d8754z-02046d111ec27d57-1---d8754z-;received=192.168.1.2;rport=35288
From: "asterisk"<sip:1000@192.168.1.3>;tag=27dad404
To: "asterisk"<sip:1000@192.168.1.3>;tag=as04683016
Call-ID: NjY4MzBlYmYzZmFmNzcyYzVmMzk4OTIwM2Q5OWMxYWI.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Jun 13 22:46:03] NOTICE[9327]: chan_sip.c:15094 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog 'NjY4MzBlYmYzZmFmNzcyYzVmMzk4OTIwM2Q5OWMxYWI.' Method: SUBSCRIBE

<--- SIP read from 62.4.81.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK53be5e44;rport
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>;tag=as72983c3d
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 103 INVITE
Contact: <sip:34672523701@62.4.81.180>
user-agent: SIP gate
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length:   212

v=0
o=root 14978 14978 IN IP4 192.168.1.1
s=session
c=IN IP4 192.168.1.1
t=0 0
m=audio 7072 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.1:7072
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x402 (gsm|ilbc), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.1:7072
list_route: hop: <sip:siproxd@192.168.1.1:5060;lr>
set_destination: Parsing <sip:siproxd@192.168.1.1:5060;lr> for address/port to send to
set_destination: set destination to 192.168.1.1, port 5060
Transmitting (no NAT) to 192.168.1.1:5060:
ACK sip:34672523701@62.4.81.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3613b1e2;rport
Route: <sip:siproxd@192.168.1.1:5060;lr>
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>;tag=as72983c3d
Contact: <sip:463910@192.168.1.3>
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/voovox-outbound-094a92e0 answered IAX2/voovox-7174
Really destroying SIP dialog '2eff1cba4119999760be29ab4399f39e@127.0.1.1' Method: REGISTER
Scheduling destruction of SIP dialog '66e61a281105676a4d0dd5bd538fd3db@192.168.1.3' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:siproxd@192.168.1.1:5060;lr> for address/port to send to
set_destination: set destination to 192.168.1.1, port 5060
Reliably Transmitting (no NAT) to 192.168.1.1:5060:
BYE sip:34672523701@62.4.81.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK09cafc31;rport
Route: <sip:siproxd@192.168.1.1:5060;lr>
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>;tag=as72983c3d
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="463910", realm="eugw.ast.voovox.com", algorithm=MD5, uri="sip:34672523701@62.4.81.180", nonce="0db80fa1", response="62987223062e09eb919b688f191e488d"
Content-Length: 0


---
  == Spawn extension (softphone, 222, 1) exited non-zero on 'IAX2/voovox-7174'
    -- Hungup 'IAX2/voovox-7174'

<--- SIP read from 62.4.81.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK09cafc31;rport
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: "14805882534" <sip:463910@192.168.1.3>;tag=as5762cc47
To: <sip:34672523701@eugw.ast.voovox.com>;tag=as72983c3d
Call-ID: 66e61a281105676a4d0dd5bd538fd3db@192.168.1.3
CSeq: 104 BYE
Contact: <sip:34672523701@62.4.81.180>
user-agent: SIP gate
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '66e61a281105676a4d0dd5bd538fd3db@192.168.1.3' Method: INVITE