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 024312648651@127.0.1.1 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:red5sip_user@127.0.0.1>;tag=z9hG4bK74877027 To: <sip:40016@127.0.0.1>;tag=as759e6d0c Call-ID: 397099427934@127.0.1.1 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK0858799 Max-Forwards: 70 To: <sip:40016@127.0.0.1> From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 CSeq: 1 ACK Contact: <sip:red5sip_user@127.0.1.1:5070> 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK19482100 Max-Forwards: 70 To: <sip:40016@127.0.0.1> From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 CSeq: 2 INVITE Contact: <sip:red5sip_user@127.0.1.1:5070> Expires: 3600 User-Agent: mjsip stack 1.6 Authorization: Digest username="red5sip_user", realm="asterisk", nonce="308fba53", uri="sip:40016@127.0.0.1", 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 - 263147788729@127.0.1.1 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:red5sip_user@127.0.1.1:5070> <--- 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:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 To: <sip:40016@127.0.0.1> Call-ID: 263147788729@127.0.1.1 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:40016@127.0.0.1:5060> Content-Length: 0 <------------> <--- SIP read from UDP:127.0.0.1:5070 ---> ACK sip:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398 Max-Forwards: 70 To: <sip:40016@127.0.0.1>;tag=as6bc959f2 From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 CSeq: 1 ACK User-Agent: mjsip stack 1.6 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '263147788729@127.0.1.1' 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:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 To: <sip:40016@127.0.0.1>;tag=as69b73555 Call-ID: 263147788729@127.0.1.1 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398 Max-Forwards: 70 To: <sip:40016@127.0.0.1>;tag=as6bc959f2 From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398 Max-Forwards: 70 To: <sip:40016@127.0.0.1>;tag=as6bc959f2 From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398 Max-Forwards: 70 To: <sip:40016@127.0.0.1>;tag=as6bc959f2 From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK47244101 Max-Forwards: 70 To: <sip:40016@127.0.0.1> From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK29134357 Call-ID: 263147788729@127.0.1.1 CSeq: 1 ACK Contact: <sip:red5sip_user@127.0.1.1:5070> 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:40016@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK45513102 Max-Forwards: 70 To: <sip:40016@127.0.0.1> From: "red5sip_user" <sip:red5sip_user@127.0.0.1>;tag=z9hG4bK16124726 Call-ID: 725338205557@127.0.1.1 CSeq: 1 INVITE Contact: <sip:red5sip_user@127.0.1.1:5070> 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 - 725338205557@127.0.1.1 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:red5sip_user@127.0.0.1>;tag=z9hG4bK16124726 To: <sip:40016@127.0.0.1>;tag=as65c60e65 Call-ID: 725338205557@127.0.1.1 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 '725338205557@127.0.1.1' 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:red5sip_user@127.0.0.1>;tag=z9hG4bK79605539 To: <sip:40016@127.0.0.1>;tag=as4a2fdbcb Call-ID: 400642563986@127.0.1.1 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:solomax...@gmail.com] 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 <horace.mi...@myit-solutions.com> 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:solomax...@gmail.com] 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 <horace.mi...@myit-solutions.com> 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:solomax...@gmail.com] 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 <horace.mi...@myit-solutions.com> 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%23@127.0.0.1> 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:solomax...@gmail.com] 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" <mkraju...@gmail.com> 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