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Horace Miles commented on OPENMEETINGS-2737: -------------------------------------------- Maxim, I followed the instructions to the and I am still experiencing the same issue. Here is an output from asterisk confbridge list: meetings*CLI> confbridge list Conference Bridge Name Users Marked Locked Muted ================================ ====== ====== ====== ===== 40011 1 1 No No meetings*CLI> confbridge list 40011 meetings*CLI> confbridge list 40011 Channel Flags User Profile Bridge Profile Menu CallerID ============================== ====== ================ ================ ================ ================ PJSIP/horace-tablet-00000000 MWE sip_user default_bridge default_menu horace-tablet meetings*CLI> As you will notice omsip-user is not listed as being in the conference. OMSIP-USER should be conference leader according to confbridge.conf [omsip_user] type=user marked=yes dsp_drop_silence=yes denoise=true Does omsip-user show as being a member of your conference call? > 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? -- This message was sent by Atlassian Jira (v8.20.7#820007)