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
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <[email protected]> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am
assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a
setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to
misconfiguration. Unfortunately this integration is not simple by design :( I'm
using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in
openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have
password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <[email protected]> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was
that Openmeetings to be access from the internet needed to be on a public
address. That address would be the one in the config.xml. If I a mistaken let
me know.
Can I have your thoughts on the following:
I am unable to get the sip agent to bind to 127.0.0.1. It refuses to bind
unless I have bind it to the same address that is in red5home
/webapps/openmeetings/public/config.xml
The problem appears to be either that the SIP protocol wants to use 127.0.0.1
for the subscribe or invites and SIP agent is bound to the Public IP address.
Therefore it is generating the error for seqno 2 which would be the SIP Invite
(I am assuming). I have not been able to get the SIP tansport to bind to
127.0.0.1 which would probably solve this problem.
Your thoughts/
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Friday, July 25, 2014 7:22 AM
To: Horace Miles
Subject: Re: VOIP and Sip Integration
hope you will be able to fix it, please let ne know if additional help is
required
On 25 July 2014 20:53, Horace Miles <[email protected]> wrote:
Hey thanks for the files.
I compared and I have found the following:
It appears the integration is setup for for a box that is NAT’ed. I thought
openmeetings had to be on a static public IP address?
So I changed every place that is referencing 127.0.0.1 to my IP address.
The Sip Agent/Openmeetings Manager does not come into the room until I restart
Asterisk. I can see it successfully logging on and then immediately logging
off. The room is successfully spawned.
There seem to be a problem with the manager once it signs on with the sip
handshake (again I am guessing)
chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission
#########@127.0.0.1 <mailto:%23#%23%23%23%23%23%23%[email protected]> for seqno 2
(Critical Response) see…… Packet timed out afer 32000ms with no response.
I will load wireshark later today on the PBX to see what else I might find.
Thanks for all your help.
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Thursday, July 24, 2014 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <[email protected]> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on
Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
