I am still trying to integrate RED5SIP and VOIP into Openmeetings 3.0 The
connection is be Declined as not authorized but I can not figure out why. Here
are the relative log and debug files. Hopefully someone can help me figure
this out.
Thanks Miles
Asterisk messages log
Aug 5 06:08:51] Asterisk 11.11.0 built by root @ vms on a i686 running Linux
on 2014-07-26 19:25:45 UTC
[Aug 5 06:08:51] NOTICE[5128] loader.c: 2 modules will be loaded.
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Connecting asterisk
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc: Connected to asterisk
[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Registered ODBC class 'asterisk'
dsn->[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Connecting mysql2
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc: Connected to mysql2
[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Registered ODBC class 'mysql2'
dsn->[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc loaded.
[Aug 5 06:08:51] NOTICE[5128] config.c: Registered Config Engine odbc
[Aug 5 06:08:51] NOTICE[5128] cdr.c: CDR simple logging enabled.
[Aug 5 06:08:51] NOTICE[5128] loader.c: 201 modules will be loaded.
[Aug 5 06:08:51] NOTICE[5128] res_smdi.c: No SMDI interfaces are available to
listen on, not starting SMDI listener.
[Aug 5 06:08:51] NOTICE[5128] config.c: Registered Config Engine sqlite3
[Aug 5 06:08:52] NOTICE[5128] chan_skinny.c: Configuring skinny from
skinny.conf
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to
'userbase' (on reload) at line 23.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to
'vmsecret' (on reload) at line 31.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'hassip'
(on reload) at line 35.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'hasiax'
(on reload) at line 39.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to
'hasmanager' (on reload) at line 47.
[Aug 5 06:08:52] NOTICE[5128] cel_custom.c: No mappings found in
cel_custom.conf. Not logging CEL to custom CSVs.
[Aug 5 06:08:52] WARNING[5128] pbx.c: Extension '_400X!' priority 5 in
'rooms', label 'ok' already in use at priority 2
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: Starting AEL load process.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: parsed config file
name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: checked config file
name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: compiled config
file name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: merged config file
name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: verified config
file name '/etc/asterisk/extensions.ael'.
/var/log/asterisk# netstat -tlvn
Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address Foreign Address State
tcp 0 0 127.0.0.1:3306 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:1935 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:9999 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN
tcp 0 0 127.0.0.1:53 0.0.0.0:* LISTEN
tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:1720 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:5080 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:25 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:39806 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN
tcp6 0 0 :::80 :::* LISTEN
tcp6 0 0 ::1:631 :::* LISTEN
tcp6 0 0 :::25 :::* LISTEN
asterisk -rvvvvvv
Connected to Asterisk 11.11.0 currently running on vms (pid = 5128)
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [40016@rooms-red5sip:1] GotoIf("SIP/red5sip_user-00000005",
"0?ok:notavail") in new stack
-- Goto (rooms-red5sip,40016,3)
-- Executing [40016@rooms-red5sip:3] Hangup("SIP/red5sip_user-00000005",
"") in new stack
== Spawn extension (rooms-red5sip, 40016, 3) exited non-zero on
'SIP/red5sip_user-00000005'
[Aug 5 06:14:19] WARNING[5164]: chan_sip.c:4175 retrans_pkt: Retransmission
timeout reached on transmission [email protected] for seqno 2 (Critical
Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
sip debug logs
CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:127.0.0.1:5070 --->
<------------->
Retransmitting #7 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
127.0.1.1:5070;branch=z9hG4bK5224484;received=127.0.0.1;rport=5070
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK74877027
To: <sip:[email protected]>;tag=as759e6d0c
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK0858799
Max-Forwards: 70
To: <sip:[email protected]>
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 1 ACK
Contact: <sip:[email protected]:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK19482100
Max-Forwards: 70
To: <sip:[email protected]>
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Authorization: Digest username="red5sip_user", realm="asterisk",
nonce="308fba53", uri="sip:[email protected]", algorithm=MD5,
response="9a2776ea6883adb1345d50eb1fed5d45"
Content-Length: 324
Content-Type: application/sdp
v=0
o=red5sip_user 0 0 IN IP4 127.0.1.1
s=Session SIP/SDP
c=IN IP4 127.0.1.1
t=0 0
m=audio 3010 RTP/AVP 8 18 0 111
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:111 ILBC/8000/1
a=fmtp:111 mode=30
a=ptime:20
m=video 7010 RTP/AVP 35
a=rtpmap:35 H264/90000/1
<------------->
--- (13 headers 15 lines) ---
Sending to 127.0.0.1:5070 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'red5sip_user' for 'red5sip_user' from 127.0.0.1:5070
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 111
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format ILBC for ID 111
Found RTP video format 35
Found video description format H264 for ID 35
Capabilities: us - (ulaw|h264), peer -
audio=(ulaw|alaw|g729|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.1.1:3010
Peer video RTP is at port 127.0.1.1:7010
Looking for 40016 in rooms-red5sip (domain 127.0.0.1)
list_route: hop: <sip:[email protected]:5070>
<--- Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
127.0.1.1:5070;branch=z9hG4bK19482100;received=127.0.0.1;rport=5070
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6bc959f2
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms
(Method: INVITE)
<--- Reliably Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
127.0.1.1:5070;branch=z9hG4bK19482100;received=127.0.0.1;rport=5070
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
To: <sip:[email protected]>;tag=as69b73555
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6bc959f2
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6bc959f2
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6bc959f2
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK47244101
Max-Forwards: 70
To: <sip:[email protected]>
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK29134357
Call-ID: [email protected]
CSeq: 1 ACK
Contact: <sip:[email protected]:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK45513102
Max-Forwards: 70
To: <sip:[email protected]>
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK16124726
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 324
Content-Type: application/sdp
v=0
o=red5sip_user 0 0 IN IP4 127.0.1.1
s=Session SIP/SDP
c=IN IP4 127.0.1.1
t=0 0
m=audio 3010 RTP/AVP 8 18 0 111
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:111 ILBC/8000/1
a=fmtp:111 mode=30
a=ptime:20
m=video 7010 RTP/AVP 35
a=rtpmap:35 H264/90000/1
<------------->
--- (12 headers 15 lines) ---
Sending to 127.0.0.1:5070 (NAT)
Sending to 127.0.0.1:5070 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'red5sip_user' for 'red5sip_user' from 127.0.0.1:5070
<--- Reliably Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.1.1:5070;branch=z9hG4bK45513102;received=127.0.0.1;rport=5070
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK16124726
To: <sip:[email protected]>;tag=as65c60e65
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c80ccd0"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms
(Method: INVITE)
Retransmitting #6 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
127.0.1.1:5070;branch=z9hG4bK4721088;received=127.0.0.1;rport=5070
From: "red5sip_user" <sip:[email protected]>;tag=z9hG4bK79605539
To: <sip:[email protected]>;tag=as4a2fdbcb
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #1 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
