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https://issues.apache.org/jira/browse/OPENMEETINGS-2737?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=17538233#comment-17538233
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Maxim Solodovnik commented on OPENMEETINGS-2737:
------------------------------------------------
Hello [~hormiles],
sorry I'm a bit slow due do kids and day-time-job :(
I can't provide any fix for version {{6.2.0}} it is unchangeable since released
so {{Affects Version/s == 6.2.0}}
and {{Fix Version/s == TBD}}
According to your issue with reloading:
{code}
Failed to bind to 0.0.0.0:5060: Address already in use
{code}
I would check
# how many asterisk processes do you have
# what other service might be listening at port 5060
According to calls from room:
previously there was section in the docs:
https://github.com/apache/openmeetings/blob/4.0.x/openmeetings-server/src/site/xdoc/voip-sip-integration.xml#L190
{code}
[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
{code}
Call from land-line/mobile to room should be available
I'll ask colleague of mine if he'll remember how it was done
Call between conf rooms was never implemented, most probably it is possible
(not sure yet, how :(( )
I can only test at my local machine
So I would appreciate any help on how can I configure local Asterisk to test
external calls :)
> 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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