rendered paste body
<--- Reliably Transmitting (no NAT) to 10.4.0.10:62206 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-4b10a0029278c628-1--d87543-;received=10.4.0.10;rport=62206
From: "401"<sip:401@asterisk>;tag=1f50f621
To: "202"<sip:202@asterisk>;tag=as5d7026db
Call-ID: NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1871d497"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.' in 32000 ms (Method: INVITE)
Found user '401'
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
ACK sip:202@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-4b10a0029278c628-1--d87543-;rport
To: "202"<sip:202@asterisk>;tag=as5d7026db
From: "401"<sip:401@asterisk>;tag=1f50f621
Call-ID: NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
INVITE sip:202@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-a119e36ea7561461-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:401@10.4.0.10:62206>
To: "202"<sip:202@asterisk>
From: "401"<sip:401@asterisk>;tag=1f50f621
Call-ID: NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="401",realm="asterisk",nonce="1871d497",uri="sip:202@asterisk",response="5f3ad10a00c46713be0d621f6db8e8bd",algorithm=MD5
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 457
v=0
o=- 2 2 IN IP4 10.4.0.10
s=CounterPath X-Lite 3.0
c=IN IP4 10.4.0.10
t=0 0
m=audio 38352 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 2 : y/Y3I/h5 KHJFt3Ok 10.3.0.2 38352
a=alt:2 1 : IgLNEeF8 WixFrhcp 10.4.0.10 38352
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 17 lines) ---
Sending to 10.4.0.10 : 62206 (NAT)
Using INVITE request as basis request - NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
Found user '401'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.4.0.10:38352
Got unsupported a:fmtp in SDP offer
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x60c (ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.4.0.10:38352
Looking for 202 in default (domain asterisk)
list_route: hop: <sip:401@10.4.0.10:62206>
<--- Transmitting (no NAT) to 10.4.0.10:62206 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-a119e36ea7561461-1--d87543-;received=10.4.0.10;rport=62206
From: "401"<sip:401@asterisk>;tag=1f50f621
To: "202"<sip:202@asterisk>
Call-ID: NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202@10.9.0.17>
Content-Length: 0
<------------>
-- Executing [202@default:1] NoOp("SIP/401-08670000", "") in new stack
-- Executing [202@default:2] Dial("SIP/401-08670000", "SIP/sr/202| 30") in new stack
-- Called sr/202
[Mar 4 21:32:35] WARNING[29354]: chan_sip.c:12188 handle_response_invite: Received response: "Forbidden" from '"401" <sip:401@10.9.0.17>;tag=as77880afa'
-- SIP/sr-08688000 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [202@default:3] Hangup("SIP/401-08670000", "") in new stack
== Spawn extension (default, 202, 3) exited non-zero on 'SIP/401-08670000'
Scheduling destruction of SIP dialog 'NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 10.4.0.10:62206 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-a119e36ea7561461-1--d87543-;received=10.4.0.10;rport=62206
From: "401"<sip:401@asterisk>;tag=1f50f621
To: "202"<sip:202@asterisk>;tag=as123e8038
Call-ID: NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
ACK sip:202@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-a119e36ea7561461-1--d87543-;rport
To: "202"<sip:202@asterisk>;tag=as123e8038
From: "401"<sip:401@asterisk>;tag=1f50f621
Call-ID: NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '7be30cca2aeb77b758c136cc41659d2e@10.9.0.17' Method: INVITE
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
SUBSCRIBE sip:401@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-053e9a589801fe71-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:401@10.4.0.10:62206>
To: "401"<sip:401@asterisk>
From: "401"<sip:401@asterisk>;tag=705a6664
Call-ID: YzJmYWRmNzUxNWVjYzk5MTFjNTBlMjk5MGUwMjUzNTQ.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Event: message-summary
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 10.4.0.10 : 62206 (NAT)
Found peer '401'
<--- Transmitting (no NAT) to 10.4.0.10:62206 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-053e9a589801fe71-1--d87543-;received=10.4.0.10;rport=62206
From: "401"<sip:401@asterisk>;tag=705a6664
To: "401"<sip:401@asterisk>;tag=as7f435f4e
Call-ID: YzJmYWRmNzUxNWVjYzk5MTFjNTBlMjk5MGUwMjUzNTQ.
Seq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c018835"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'YzJmYWRmNzUxNWVjYzk5MTFjNTBlMjk5MGUwMjUzNTQ.' in 32000 ms (Method: SUBSCRIBE)
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
SUBSCRIBE sip:401@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-5a252d4acd7f2252-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:401@10.4.0.10:62206>
To: "401"<sip:401@asterisk>
From: "401"<sip:401@asterisk>;tag=705a6664
Call-ID: YzJmYWRmNzUxNWVjYzk5MTFjNTBlMjk5MGUwMjUzNTQ.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="401",realm="asterisk",nonce="0c018835",uri="sip:401@asterisk",response="16febc9e3be03fa64bdf967c08675ba5",algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 10.4.0.10 : 62206 (NAT)
Found peer '401'
Looking for 401 in default (domain asterisk)
<--- Transmitting (no NAT) to 10.4.0.10:62206 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.4.0.10:62206;branch=z9hG4bK-d87543-5a252d4acd7f2252-1--d87543-;received=10.4.0.10;rport=62206
From: "401"<sip:401@asterisk>;tag=705a6664
To: "401"<sip:401@asterisk>;tag=as7f435f4e
Call-ID: YzJmYWRmNzUxNWVjYzk5MTFjNTBlMjk5MGUwMjUzNTQ.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Really destroying SIP dialog 'YzJmYWRmNzUxNWVjYzk5MTFjNTBlMjk5MGUwMjUzNTQ.' Method: SUBSCRIBE
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
<------------->
--- (0 headers 1 lines) ---
Really destroying SIP dialog 'NmNhOTQ2ODkyMjAzOGM2MzY0NDUyYjdkMzRlMzRiMDg.' Method: ACK
ch*CLI>
<--- SIP read from 10.4.0.10:62206 --->
<------------->
--- (0 headers 1 lines) ---
ch*CLI>