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https://issues.apache.org/jira/browse/OPENMEETINGS-2737?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=17552547#comment-17552547
 ] 

ASF subversion and git services commented on OPENMEETINGS-2737:
---------------------------------------------------------------

Commit 6d7b933204f7ed6ebee7b7c2e9a841aac179283d in openmeetings's branch 
refs/heads/master from Maxim Solodovnik
[ https://gitbox.apache.org/repos/asf?p=openmeetings.git;h=6d7b93320 ]

[OPENMEETINGS-2737] instructions were updated for Asterisk 18, issue in 
TimerService was fixed


> Incomplete Address when dialing OM Conference room
> --------------------------------------------------
>
>                 Key: OPENMEETINGS-2737
>                 URL: https://issues.apache.org/jira/browse/OPENMEETINGS-2737
>             Project: Openmeetings
>          Issue Type: Bug
>          Components: VoIP/SIP
>    Affects Versions: 6.2.0
>            Reporter: Horace Miles
>            Assignee: Maxim Solodovnik
>            Priority: Major
>
> When trying to call OM conference room I receive the following error:  
> SIP/2.0 484 Address Incomplete
> *CLI> pjsip show history
> No.   Timestamp  (Dir) Address                  SIP Message                   
>      
> ===== ========== ============================== 
> ===================================
> 00000 1652464465 * <== 98.174.244.227:41916     INVITE 
> sip:40011@98.174.244.232 SIP/2.0
> 00001 1652464465 * ==> 98.174.244.227:41916     SIP/2.0 401 Unauthorized
> 00002 1652464465 * <== 98.174.244.227:41916     ACK sip:40011@98.174.244.232 
> SIP/2.0
> 00003 1652464465 * <== 98.174.244.227:41916     INVITE 
> sip:40011@98.174.244.232 SIP/2.0
> 00004 1652464465 * ==> 98.174.244.227:41916     SIP/2.0 484 Address Incomplete
> 00005 1652464465 * <== 98.174.244.227:41916     ACK sip:40011@98.174.244.232 
> SIP/2.0
> *CLI>
> sip.conf settings
> [omsip_user]
> host=dynamic
> secret=<mysecret>
> context=rooms-omsip
> transport=ws,wss
> type=friend
> encryption=no
> avpf=yes
> icesupport=yes
> directmedia=no
> allow=!all,ulaw,opus,vp8
> extensions.conf configuration
> [rooms]
> exten => 
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
> exten => _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
> exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
> exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
> exten => _400X!,n,Hangup
> exten => _400X!,n(notavail),Answer()
> exten => _400X!,n,Playback(invalid)
> exten => _400X!,n,Hangup
> [rooms-originate]
> exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
> exten => _400X!,n,Hangup
> [rooms-out]
> ; *****************************************************
> ; Extensions for outgoing calls from Openmeetings room.
> ; *****************************************************
> [rooms-omsip]
> exten => 
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,omsip_user)
> exten => _400X!,n(notavail),Hangup
> Asterisk Database
> CLI> database show
> /dundi/secret                                     : 
> fL3QQ8egcjnj1bEufyh+AQ==;W6fVbQ9sJWPq0oZp50y7Ig==
> /dundi/secretexpiry                               : 1652465880               
> /openmeetings/rooms                               : 4004                     
> /openmeetings/rooms/40011                         : 7777                     
> /pbx/UUID                                         : 
> 7dd6882b-8da9-4099-a6a7-3012970c94ca
> /registrar/contact/horace-cellphone;@de16880426ac7644569b396c5df408ff: 
> {"via_addr":"10.10.0.8","qualify_timeout":"3.000000","call_id":"GM3y5EhhVO","reg_server":"","prune_on_boot":"no","path":"","endpoint":"horace-cellphone","via_port":"41916","authenticate_qualify":"no","uri":"sip:horace-cellphone@98.174.244.227:41916;transport=udp","qualify_frequency":"0","user_agent":"LinphoneAndroid/4.6.7
>  (Galaxy Note9) LinphoneSDK/5.1.28 
> (tags/5.1.28^0)","expiration_time":"1652465692","outbound_proxy":""}
> /registrar/contact/horace-desktop;@2487af86a629ea26178ed30c7963b8f8: 
> {"via_addr":"10.10.0.2","qualify_timeout":"3.000000","call_id":"2LzZJqpTs1","reg_server":"","prune_on_boot":"no","path":"","endpoint":"horace-desktop","via_port":"5060","authenticate_qualify":"no","uri":"sip:horace-desktop@98.174.244.227;transport=udp","qualify_frequency":"0","user_agent":"Linphone
>  Desktop/4.4.1 (MILES-PC) Windows 10 Version 2009, Qt 5.15.2 
> LinphoneCore/5.1.19-1-g6cdd0918e","expiration_time":"1652466228","outbound_proxy":""}
> 7 results found.
> *CLI> 
> I am using linphone 4.4.1 - Qt5.15.2
> Asterisk 16
> I can successfully make calls from Asterisk extension inbound and output to 
> both internal extentions and external PTSN numbers.
> I can not dial out of a OM Conference room - I get nothing at all
> I can not dial into a open meetings 
> I can not dial between conference rooms
> I have also tried to create AOR, Auth and Endpoint records for a conference 
> room as follows:
> [40011]
> type=endpoint
> context=rooms-omsip
> disallow=all
> allow=ulaw
> auth=4011-auth
> aors=40011
> [40011-auth]
> type=auth
> auth_type=userpass
> username=40011
> password=<somepassword>
> [40011]
> type=aor
> max_contacts=25
> With the above configuration I receive the same error  484 Address incomplete
> If I change the context to something like home-phones, I receive the 
> following error:
> *CLI>   == Setting global variable 'SIPDOMAIN' to '98.174.244.232'
>     -- Executing [40011@home-phones:1] 
> Dial("PJSIP/horace-cellphone-00000001", "PJSIP/40011") in new stack
> [May 13 11:19:01] ERROR[4701]: res_pjsip.c:3562 ast_sip_create_dialog_uac: 
> Endpoint '40011': Could not create dialog to invalid URI '40011'.  Is 
> endpoint registered and reachable?
> [May 13 11:19:01] ERROR[4701]: chan_pjsip.c:2687 request: Failed to create 
> outgoing session to endpoint '40011'
> [May 13 11:19:01] WARNING[4734][C-00000002]: app_dial.c:2576 dial_exec_full: 
> Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
>     -- No devices or endpoints to dial (technology/resource)
>     -- Auto fallthrough, channel 'PJSIP/horace-cellphone-00000001' status is 
> 'CHANUNAVAIL'
> Can you help me to figure this out to be able to call into a conference room 
> from external number and to be able to call conf->conf and conf-external?



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