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https://issues.apache.org/jira/browse/OPENMEETINGS-2737?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=17537545#comment-17537545
]
Horace Miles commented on OPENMEETINGS-2737:
--------------------------------------------
Good morning Maxim,
Thanks for your response. I used your steps/guide for Asteriskintegrations. I
have confirmed all setting. I dialed into the conference room and there was
silence no error codes. But since I didn't know what to expected I want to do
a capture. I restarted asterisk and it no longer works. At this point I am
ready to pay for commercial support. This should not be this hard. I can do
everything I want with OM and asterisk except these couple of things.
/etc/asterisk/res_config_mysql.conf:
[general]
dbhost = 127.0.0.1
dbname = openmeetings
dbuser = root
dbpass = <mypassword>
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
dbcharset = utf8
requirements=warn ; or createclose or createchar
meetings*CLI> sip reload
Reloading SIP
[May 16 05:44:37] WARNING[1416]: chan_sip.c:33496 reload_config: Failed to bind
to 0.0.0.0:5060: Address already in use
meetings*CLI>
meetings*CLI> database show
/openmeetings/rooms/40011 : 1234
/openmeetings/rooms/4004 : 1234
/
meetings*CLI> dialplan reload
Dialplan reloaded.
== Setting global variable 'CONSOLE' to 'Console/dsp'
== Setting global variable 'IAXINFO' to 'guest'
== Setting global variable 'TRUNK' to 'DAHDI/G2'
== Setting global variable 'TRUNKMSD' to '1'
-- Including switch 'DUNDi/e164' in context 'dundi-e164-switch'
[May 16 06:08:17] WARNING[2225]: pbx.c:7123 add_priority: Extension '_400X!'
priority 5 in 'rooms', label 'ok' already in use at priority 2
[May 16 06:08:17] WARNING[2225]: pbx.c:7156 add_priority: Unable to register
extension '_400X!' priority 1 in 'rooms-omsip', already in use
[May 16 06:08:17] WARNING[2225]: pbx_config.c:1891 pbx_load_config: Unable to
register extension at line 906 of extensions.conf
[May 16 06:08:17] WARNING[2225]: pbx.c:7156 add_priority: Unable to register
extension '_400X!' priority 2 in 'rooms-omsip', already in use
[May 16 06:08:17] WARNING[2225]: pbx_config.c:1891 pbx_load_config: Unable to
register extension at line 907 of extensions.conf
[May 16 06:08:17] WARNING[2225]: pbx.c:7156 add_priority: Unable to register
extension '_400X!' priority 3 in 'rooms-omsip', already in use
[May 16 06:08:17] WARNING[2225]: pbx_config.c:1891 pbx_load_config: Unable to
register extension at line 908 of extensions.conf
-- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
-- Including switch 'Lua/' in context 'default'
-- Including switch 'Lua/' in context 'public'
-- Including switch 'Lua/' in context 'demo'
-- Including switch 'Lua/' in context 'local'
-- Time to scan old dialplan and merge leftovers back into the new:
0.003524 sec
-- Time to restore hints and swap in new dialplan: 0.000010 sec
-- Time to delete the old dialplan: 0.000186 sec
-- Total time merge_contexts_delete: 0.003720 sec
-- pbx_config successfully loaded 57 contexts (enable debug for details).
meetings*CLI> sip reload
Reloading SIP
[May 16 06:08:22] WARNING[1416]: chan_sip.c:33496 reload_config: Failed to bind
to 0.0.0.0:5060: Address already in use
meetings*CLI> pjsip reload
Module 'res_pjsip.so' reloaded successfully.
Module 'res_pjsip_authenticator_digest.so' reloaded successfully.
Module 'res_pjsip_endpoint_identifier_ip.so' reloaded successfully.
Module 'res_pjsip_mwi.so' reloaded successfully.
Module 'res_pjsip_notify.so' reloaded successfully.
Module 'res_pjsip_outbound_publish.so' reloaded successfully.
Module 'res_pjsip_publish_asterisk.so' reloaded successfully.
Module 'res_pjsip_outbound_registration.so' reloaded successfully.
-- Reloading module 'res_pjsip.so' (Basic SIP resource)
[May 16 06:08:26] NOTICE[3333]: sorcery.c:1345 sorcery_object_load: Type
'system' is not reloadable, maintaining previous values
-- Reloading module 'res_pjsip_authenticator_digest.so' (PJSIP
authentication resource)
-- Reloading module 'res_pjsip_endpoint_identifier_ip.so' (PJSIP IP
endpoint identifier)
-- Reloading module 'res_pjsip_mwi.so' (PJSIP MWI resource)
-- Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)
-- Reloading module 'res_pjsip_outbound_publish.so' (PJSIP Outbound Publish
Support)
-- Reloading module 'res_pjsip_publish_asterisk.so' (PJSIP Asterisk Event
PUBLISH Support)
-- Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound
Registration Support)
meetings*CLI>
I ran executed
cd /etc/asterisk
sudo
/usr/src/asterisk/asterisk-16.13.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
and it converted the pjsip to look this below: (Usernames/Passwords redacted)
/
;--
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements start
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
[general]
allowoverlap = no
[omsip_user]
transport = ws,wss
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
--;
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
[omsip_user]
type = aor
max_contacts = 1
maximum_expiration = 43200
[omsip_user]
type = auth
username = omsip_user
password = password
[omsip_user]
type = endpoint
context = rooms-omsip
allow = !all,ulaw,opus,vp8
force_rport = yes
rewrite_contact = yes
ice_support = yes
direct_media = no
use_avpf = yes
auth = omsip_user
outbound_auth = omsip_user
aors = omsip_user
[voipms]
type = registration
transport = transport-udp
outbound_auth = voipms
client_uri = sip:[email protected]:5060 ; (one of our multiple
servers, you can choose the one closer to your location)
server_uri = sip:sanjose2.voip.ms:5060 ; (one of our multiple
servers, you can choose the one closer to your location)
[voipms]
type = auth
auth_type = userpass
username = voipmsuserid ; (Replace with your 6 digit Main SIP
Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
password = voipmspassword ; your password
[voipms]
type = aor
contact = sip:[email protected] ; (one of our multiple
servers, you can choose the one closer to your location)
[voipms]
type = endpoint
transport = transport-udp
context = mycontext
disallow = all
allow = ulaw
; allow=g729 ; uncomment if you support g729
from_user = voipmsuserid ; (Replace with your 6 digit Main SIP
Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
auth = voipms
outbound_auth = voipms
aors = voipms
; NAT parameters:
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
[voipms]
type = identify
endpoint = voipms
match = sanjose2.voip.ms ; (one of our multiple servers, you can choose
the one closer to your location)
[horace-cellphone]
type=endpoint
context=home-phones
disallow=all
allow=ulaw
auth=horace-cellphone-auth
aors=horace-cellphone
[horace-cellphone-auth]
type=auth
auth_type=userpass
username=horace-cellphone
password=Peoria!2#4
[horace-cellphone]
type=aor
max_contacts=1
[horace-desktop]
type=endpoint
context=home-phones
disallow=all
allow=ulaw
auth=horace-desktop-auth
aors=horace-desktop
[horace-desktop-auth]
type=auth
auth_type=userpass
username=horace-desktop
password=Peoria!2#4
[horace-desktop]
type=aor
max_contacts=1
/
********************************
out put from commands
meetings*CLI> pjsip show history entry 2
<--- History Entry 2 Received from 98.174.244.227:41916 at 1652707615 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
10.10.0.8:41916;rport=41916;received=98.174.244.227;branch=z9hG4bK.lDoGywsrz
Call-ID: aUMs9ubixh
From: "Horace Cellphone" <sip:[email protected]>;tag=FMoo-rNPO
To: <sip:[email protected]>;tag=z9hG4bK.lDoGywsrz
Contact:
<sip:[email protected]:41916;transport=udp>;expires=3599;+sip.instance="<urn:uuid:b8fc657a-451d-00f9-b74d-6a441ea04ea8>"
Max-Forwards: 70
CSeq: 20 ACK
Content-Length: 0
meetings*CLI> pjsip show history entry 3
<--- History Entry 3 Received from 98.174.244.227:41916 at 1652707615 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
10.10.0.8:41916;rport=41916;received=98.174.244.227;branch=z9hG4bK.4zwPNeUcC
From: "Horace Cellphone" <sip:[email protected]>;tag=FMoo-rNPO
To: <sip:[email protected]>
CSeq: 21 INVITE
Call-ID: aUMs9ubixh
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 523
Contact:
<sip:[email protected]:41916;transport=udp>;expires=3599;+sip.instance="<urn:uuid:b8fc657a-451d-00f9-b74d-6a441ea04ea8>"
User-Agent: LinphoneAndroid/4.6.7 (Galaxy Note9) LinphoneSDK/5.1.28
(tags/5.1.28^0)
Authorization: Digest username="horace-cellphone", realm="asterisk",
nonce="1652707615/79757b2db41d6dd9517c2aea645cb174",
uri="sip:[email protected]", response="af7b6c161bd567de2708c47aea0fd4a9",
algorithm=md5, cnonce="1C1ZVQJV9rzgHEVA", opaque="723fe28a362e5f59", qop=auth,
nc=00000001
Content-Type: application/sdp
Content-Length: 523
v=0
o=horace-cellphone 478 2088 IN IP4 10.10.0.8
s=Talk
c=IN IP4 10.10.0.8
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 18 99 100 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:99 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:101 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
meetings*CLI> pjsip show history entry 4
<--- History Entry 4 Sent to 98.174.244.227:41916 at 1652707615 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
10.10.0.8:41916;rport=41916;received=98.174.244.227;branch=z9hG4bK.4zwPNeUcC
Call-ID: aUMs9ubixh
From: "Horace Cellphone" <sip:[email protected]>;tag=FMoo-rNPO
To: <sip:[email protected]>;tag=8a0b21c5-2789-4bcb-887e-cf1d236ae39b
CSeq: 21 INVITE
Server: Asterisk PBX 16.13.0
Content-Length: 0
meetings*CLI> 5
No such command '5' (type 'core show help 5' for other possible commands)
meetings*CLI> pjsip show history entry 5
<--- History Entry 5 Received from 98.174.244.227:41916 at 1652707615 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
10.10.0.8:41916;rport=41916;received=98.174.244.227;branch=z9hG4bK.4zwPNeUcC
Call-ID: aUMs9ubixh
From: "Horace Cellphone" <sip:[email protected]>;tag=FMoo-rNPO
To: <sip:[email protected]>;tag=8a0b21c5-2789-4bcb-887e-cf1d236ae39b
Contact:
<sip:[email protected]:41916;transport=udp>;expires=3599;+sip.instance="<urn:uuid:b8fc657a-451d-00f9-b74d-6a441ea04ea8>"
Max-Forwards: 70
CSeq: 21 ACK
Content-Length: 0
meetings*CLI>
> 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
> Fix For: 6.2.0
>
>
> 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:[email protected] SIP/2.0
> 00001 1652464465 * ==> 98.174.244.227:41916 SIP/2.0 401 Unauthorized
> 00002 1652464465 * <== 98.174.244.227:41916 ACK sip:[email protected]
> SIP/2.0
> 00003 1652464465 * <== 98.174.244.227:41916 INVITE
> sip:[email protected] 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:[email protected]
> 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:[email protected]: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:[email protected];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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