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https://issues.apache.org/jira/browse/OPENMEETINGS-2737?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=17552461#comment-17552461
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Horace Miles commented on OPENMEETINGS-2737:
--------------------------------------------
Maxim,
I have figured out why I was getting the bad address and why I couldn't dial
into the conferences.
1. In extensions.conf I needed to add include rooms and rooms-omsip contexts
to my home-phones context. This allowed me to dial into the conference rooms
2. After dialing into the conferences I was receiving one way audio, i.e. I
was not recieving any of the responses from asterisk at the client that dialed
in. Was not receiving the prompt to enter the PIN for the conference room.
When I would enter the PIN for the conference because I knew it was asking for
it, I was able to connect to the conference but still would not receive any
audio back. This problem was caused by the Linphone client for android.
a.) Linphone client needs to be configured to use a STUN server. The only
way to do that on the Android client is to have the client download a
configuration file. I switched to MizuDroid client and I can dial into the
conference and hear the system prompts.
Now the problem I am having is Openmeetings/Asterisk is not recognizing when
the moderatator enters the conference room. So the caller told the conference
will start when the Leader joins the conference. It never recognizes when the
moderator joins and so they are just stuck with music on hold. Can you look
into that?
Miles
> 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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