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https://issues.apache.org/jira/browse/OPENMEETINGS-2737?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=17554265#comment-17554265
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Horace Miles commented on OPENMEETINGS-2737:
--------------------------------------------
HI MAXIM,
Thanks
1 and 2 make sense, but should valid correction params/credentials already
exist in the file?
I got the information for tomcat4 from this jira
https://issues.apache.org/jira/browse/OPENMEETINGS-2410
[^agent-confid.gsm]
Anyone can use the agent-confid.gsm file provided. However if anyone is looking
for custom gsm files, I am willing to provide them for a fee. This one is in
english.
r-agent-confid.gsm is in Russian, for you for helping me. Thanks.
I also made my trunk number the main call in number for the conference rooms, I
create agent-confid.gsm to ask for the conference ID which uses the scripting
from the rooms context to ask for the pin and passes the caller into the
conference. It will also ask the caller to announce their name and play music
on hold until the moderator enters the conference.
[^r-agent-confid.gsm]
Extensions.conf entries:
[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
exten => _XXXXXXXXXX,1,Set(CALLERID(all)="YAH's Global Kingdom Ministries
<4803829901>")
same => n,Dial(PJSIP/${EXTEN}@voipms)
include => home-phones
[voipms-inbound]
exten => <YOUR DID>,1,Answer() ;your DID
same => n,PLayback(agent-confid)
same => n,WaitExten
exten =>
_400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${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
exten=> 1,1,Answer()
same => n,Dial(PJSIP/desktop)
exten => 2,1,Answer()
same => n,Dial(PJSIP/cellphone)
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX
handler will match the incoming call and create a loop
;
include => voipms-inbound
include => voipms-outbound
include => rooms-out
;include => mainmenu
> 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
> Attachments: agent-confid.gsm, r-agent-confid.gsm
>
>
> 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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