rendered paste bodysip set debug on
SIP Debugging enabled
crowenix*CLI>
crowenix*CLI>
crowenix*CLI>
[Oct 22 16:25:07] NOTICE[16711]: chan_sip.c:11655 sip_reregister: -- Re-registration for 4083517293@cps.onvoip.net
> doing dnsmgr_lookup for 'cps.onvoip.net'
> ast_get_srv: SRV lookup for '_sip._UDP.cps.onvoip.net' mapped to host cps1.onvoip.net, port 5060
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.116.23:5060:
REGISTER sip:cps.onvoip.net SIP/2.0
Via: SIP/2.0/UDP 68.167.143.51:5060;branch=z9hG4bK7c9935ae;rport
Max-Forwards: 70
From: <sip:4083517293@cps.onvoip.net>;tag=as106c0341
To: <sip:4083517293@cps.onvoip.net>
Call-ID: 52f772153f14e466022d66fc3e37e8eb@127.0.1.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Authorization: Digest username="4083517293", realm="BroadWorks", algorithm=MD5, uri="sip:cps.onvoip.net", nonce="BroadWorksXgu38ip65Twqmv7dBW", response="4d97c4d8383bd7124e4b3a40ed6ad6e7", qop=auth, cnonce="60b3edf0", nc=00000002
Expires: 120
Contact: <sip:4083517293@68.167.143.51>
Content-Length: 0
---
<--- SIP read from UDP:204.11.116.23:32820 --->
SIP/2.0 200 OK
Via:SIP/2.0/UDP 68.167.143.51:5060;branch=z9hG4bK7c9935ae;rport
From:<sip:4083517293@cps.onvoip.net>;tag=as106c0341
To:<sip:4083517293@cps.onvoip.net>;tag=1145967996-1319325907624
Call-ID:52f772153f14e466022d66fc3e37e8eb@127.0.1.1
CSeq:104 REGISTER
Contact:<sip:4083517293@68.167.143.51>;q=0.5;expires=119
Allow-Events:call-info,line-seize,dialog,message-summary,as-feature-event
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '52f772153f14e466022d66fc3e37e8eb@127.0.1.1' in 32000 ms (Method: REGISTER)
[Oct 22 16:25:07] NOTICE[16711]: chan_sip.c:18392 handle_response_register: Outbound Registration: Expiry for cps.onvoip.net is 119 sec (Scheduling reregistration in 104 s)
<--- SIP read from UDP:68.167.143.53:5060 --->
INVITE sip:5001414@68.167.143.51:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bK50ef7ca5A6C8E47A
Record-Route: <sip:1000@68.167.143.53;lr>
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 1 INVITE
Contact: <sip:1000@68.167.143.53:5060>
User-agent: PolycomSoundPointIP-SPIP_330-UA/3.2.5.0508
Supported: 100rel
Supported: replaces
allow-events: talk
allow-events: hold
allow-events: conference
Max-forwards: 70
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Content-Type: application/sdp
Accept-Language: en
Content-Length: 226
v=0
o=- 1167611767 1167611767 IN IP4 68.167.143.53
s=Polycom IP Phone
c=IN IP4 68.167.143.53
t=0 0
a=sendrecv
m=audio 16400 RTP/AVP 0 8 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
<------------->
--- (19 headers 10 lines) ---
== Using SIP RTP CoS mark 5
Sending to 68.167.143.53 : 5060 (no NAT)
Using INVITE request as basis request - b9636f83-fa8aae10-65e3cfb1@192.168.1.150
Found peer '1000' for '1000' from 68.167.143.53:5060
<--- Reliably Transmitting (no NAT) to 68.167.143.53:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bK50ef7ca5A6C8E47A;received=68.167.143.53
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>;tag=as78054864
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d628952"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'b9636f83-fa8aae10-65e3cfb1@192.168.1.150' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:68.167.143.53:5060 --->
ACK sip:5001414@68.167.143.51:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bK50ef7ca5A6C8E47A
Record-Route: <sip:1000@68.167.143.53;lr>
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>;tag=as78054864
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 1 ACK
Contact: <sip:1000@68.167.143.53:5060>
User-agent: PolycomSoundPointIP-SPIP_330-UA/3.2.5.0508
Max-forwards: 70
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Accept-Language: en
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
<--- SIP read from UDP:68.167.143.53:5060 --->
INVITE sip:5001414@68.167.143.51:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bKb83cd8368BE088AF
Record-Route: <sip:1000@68.167.143.53;lr>
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 2 INVITE
Contact: <sip:1000@68.167.143.53:5060>
Authorization: Digest username="1000", realm="asterisk", nonce="0d628952", uri="sip:5001414@192.168.1.1:5060;user=phone", response="4d31102154eb208cd282ef8d601543fa", algorithm=MD5
User-agent: PolycomSoundPointIP-SPIP_330-UA/3.2.5.0508
Supported: 100rel
Supported: replaces
allow-events: talk
allow-events: hold
allow-events: conference
Max-forwards: 70
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Content-Type: application/sdp
Accept-Language: en
Content-Length: 226
v=0
o=- 1167611767 1167611767 IN IP4 68.167.143.53
s=Polycom IP Phone
c=IN IP4 68.167.143.53
t=0 0
a=sendrecv
m=audio 16400 RTP/AVP 0 8 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
<------------->
--- (20 headers 10 lines) ---
Sending to 68.167.143.53 : 5060 (no NAT)
Using INVITE request as basis request - b9636f83-fa8aae10-65e3cfb1@192.168.1.150
Found peer '1000' for '1000' from 68.167.143.53:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 127
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 127
Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=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 68.167.143.53:16400
Looking for 5001414 in myphones (domain 68.167.143.51)
list_route: hop: <sip:1000@68.167.143.53;lr>
<--- Transmitting (no NAT) to 68.167.143.53:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bKb83cd8368BE088AF;received=68.167.143.53
Record-Route: <sip:1000@68.167.143.53;lr>
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5001414@68.167.143.51>
Content-Length: 0
<------------>
-- Executing [5001414@myphones:1] Log("SIP/1000-00000004", "NOTICE, Dialing out from "1000" <1000> to 001414 through Foo Provider") in new stack
[Oct 22 16:25:08] NOTICE[16791]: Ext. 5001414:1 @ myphones: Dialing out from "1000" <1000> to 001414 through Foo Provider
-- Executing [5001414@myphones:2] Dial("SIP/1000-00000004", "SIP/cps.onvoip.net/001414,60") in new stack
== Using SIP RTP CoS mark 5
> ast_get_srv: SRV lookup for '_sip._UDP.cps.onvoip.net' mapped to host cps1.onvoip.net, port 5060
Audio is at 68.167.143.51 port 11756
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.116.23:5060:
INVITE sip:001414@cps.onvoip.net SIP/2.0
Via: SIP/2.0/UDP 68.167.143.51:5060;branch=z9hG4bK038436de;rport
Max-Forwards: 70
From: "1000" <sip:1000@68.167.143.51>;tag=as3b0ef58c
To: <sip:001414@cps.onvoip.net>
Contact: <sip:1000@68.167.143.51>
Call-ID: 1ac55d585aa396953b7ef51529bab32c@68.167.143.51
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Date: Sat, 22 Oct 2011 23:25:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296
v=0
o=root 1700262596 1700262596 IN IP4 68.167.143.51
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 68.167.143.51
t=0 0
m=audio 11756 RTP/AVP 0 3 8 127
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=ptime:20
a=sendrecv
---
-- Called cps.onvoip.net/001414
<--- SIP read from UDP:204.11.116.23:32820 --->
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 68.167.143.51:5060;branch=z9hG4bK038436de;rport
From:"1000"<sip:1000@68.167.143.51>;tag=as3b0ef58c
To:<sip:001414@cps.onvoip.net>
Call-ID:1ac55d585aa396953b7ef51529bab32c@68.167.143.51
CSeq:102 INVITE
Content-Length:0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:204.11.116.23:32820 --->
SIP/2.0 604 Does not exist anywhere
Via:SIP/2.0/UDP 68.167.143.51:5060;branch=z9hG4bK038436de;rport
From:"1000"<sip:1000@68.167.143.51>;tag=as3b0ef58c
To:<sip:001414@cps.onvoip.net>;tag=770414738-1319325909039
Call-ID:1ac55d585aa396953b7ef51529bab32c@68.167.143.51
CSeq:102 INVITE
Content-Length:0
<------------->
--- (7 headers 0 lines) ---
-- Got SIP response 604 "Does not exist anywhere" back from 204.11.116.23
Transmitting (no NAT) to 204.11.116.23:5060:
ACK sip:001414@cps.onvoip.net SIP/2.0
Via: SIP/2.0/UDP 68.167.143.51:5060;branch=z9hG4bK038436de;rport
Max-Forwards: 70
From: "1000" <sip:1000@68.167.143.51>;tag=as3b0ef58c
To: <sip:001414@cps.onvoip.net>;tag=770414738-1319325909039
Contact: <sip:1000@68.167.143.51>
Call-ID: 1ac55d585aa396953b7ef51529bab32c@68.167.143.51
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Content-Length: 0
---
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [5001414@myphones:3] PlayTones("SIP/1000-00000004", "congestion") in new stack
-- Executing [5001414@myphones:4] Hangup("SIP/1000-00000004", "") in new stack
== Spawn extension (myphones, 5001414, 4) exited non-zero on 'SIP/1000-00000004'
Scheduling destruction of SIP dialog 'b9636f83-fa8aae10-65e3cfb1@192.168.1.150' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 68.167.143.53:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bKb83cd8368BE088AF;received=68.167.143.53
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>;tag=as27bdb6a1
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:68.167.143.53:5060 --->
ACK sip:5001414@68.167.143.51:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.167.143.53:5060;branch=z9hG4bKb83cd8368BE088AF
Record-Route: <sip:1000@68.167.143.53;lr>
From: "1000" <sip:1000@68.167.143.51:5060>;tag=C6805F97-17BAB5F4
To: <sip:5001414@68.167.143.51:5060;user=phone>;tag=as27bdb6a1
Call-ID: b9636f83-fa8aae10-65e3cfb1@192.168.1.150
CSeq: 2 ACK
Contact: <sip:1000@68.167.143.53:5060>
User-agent: PolycomSoundPointIP-SPIP_330-UA/3.2.5.0508
Max-forwards: 70
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Accept-Language: en
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '1ac55d585aa396953b7ef51529bab32c@68.167.143.51' Method: INVITE
crowenix*CLI>