Major Description
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:
/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?