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root@Filogic-GW:/etc/asterisk# cat extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[from-internal]
exten => 1001,1,Dial(PJSIP/1001,20)
exten => 1002,1,Dial(PJSIP/1002,20)
[default]
exten => _X.,1,NoOp(Unhandled call to ${EXTEN})
exten => _X.,n,Playback(vm-nobodyavail)
exten => _X.,n,Hangup()
exten => 1001,1,Dial(PJSIP/1001,20)
exten => 1001,n,NoOp(Dial status for 1001: ${DIALSTATUS})
exten => 1001,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
exten => 1001,n(busy),Playback(all-circuits-busy-now)
exten => 1001,n(busi),Hangup()
exten => 1001,n(unavail),Playback(vm-nobodyavail)
exten => 1001,n(unavail),Hangup()
exten => 1002,1,Dial(PJSIP/1002,20)
exten => 1002,n,NoOp(Dial status for 1002: ${DIALSTATUS})
exten => 1002,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
exten => 1002,n(busy),Playback(all-circuits-busy-now)
exten => 1002,n(busy),Hangup()
exten => 1002,n(unavail),Playback(vm-nobodyavail)
exten => 1002,n(unavail),Hangup() |
5) Run the asterisk server in the board and reload both the configuration files in the asterisk CLI
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root@Filogic-GW:/etc/asterisk# asterisk -rvvv
Asterisk 18.15.1, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 18.15.1 currently running on Filogic-GW (pid = 4177657)
Filogic-GW*CLI>
Filogic-GW*CLI> module reload res_pjsip.so
Module 'res_pjsip.so' reloaded successfully.
-- Reloading module 'res_pjsip.so' (Basic SIP resource)
Filogic-GW*CLI> dialplan reload
Dialplan reloaded.
[Apr 28 17:43:14] WARNING[3460]: pbx.c:7126 add_priority: Extension '1001' priority 7 in 'default', label 'unavail' already in use at priority 6
[Apr 28 17:43:14] WARNING[3460]: pbx.c:7126 add_priority: Extension '1002' priority 5 in 'default', label 'busy' already in use at priority 4
[Apr 28 17:43:14] WARNING[3460]: pbx.c:7126 add_priority: Extension '1002' priority 7 in 'default', label 'unavail' already in use at priority 6
-- Time to scan old dialplan and merge leftovers back into the new: 0.000048 sec
-- Time to restore hints and swap in new dialplan: 0.000012 sec
-- Time to delete the old dialplan: 0.000034 sec
-- Total time merge_contexts_delete: 0.000094 sec
-- pbx_config successfully loaded 6 contexts (enable debug for details).
Filogic-GW*CLI>
|
6) Verify the clients are successfully registered with the asterisk server
5) Run the asterisk server in the board and reload both the configuration files in the asterisk CLI
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root@Filogic-GW:/etc/asterisk# asterisk -rvvv
Asterisk 18.15.1, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 18.15.1 currently running on Filogic-GW (pid = 4177657)
Filogic-GW*CLI>
Filogic-GW*CLI> module reload res_pjsip.so
Module 'res_pjsip.so' reloaded successfully.
-- Reloading module 'res_pjsip.so' (Basic SIP resource)
Filogic-GW*CLI> dialplan reload
Dialplan reloaded.
[Apr 28 17:43:14] WARNING[3460]: pbx.c:7126 add_priority: Extension '1001' priority 7 in 'default', label 'unavail' already in use at priority 6
[Apr 28 17:43:14] WARNING[3460]: pbx.c:7126 add_priority: Extension '1002' priority 5 in 'default', label 'busy' already in use at priority 4
[Apr 28 17:43:14] WARNING[3460]: pbx.c:7126 add_priority: Extension '1002' priority 7 in 'default', label 'unavail' already in use at priority 6
-- Time to scan old dialplan and merge leftovers back into the new: 0.000048 sec
-- Time to restore hints and swap in new dialplan: 0.000012 sec
-- Time to delete the old dialplan: 0.000034 sec
-- Total time merge_contexts_delete: 0.000094 sec
-- pbx_config successfully loaded 6 contexts (enable debug for details).
Filogic-GW*CLI>
|
6) If we are facing any database errors, please follow below steps
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systemctl stop asteriskcd /var/lib/asteriskmv astdb.sqlite3 /tmp/astdb.sqlite3.corruptedcorrect dialplan configuration
Restart asterisk and Re-register endpoints
systemctl start asterisk |
7) Verify the clients are successfully registered with the asterisk server
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Filogic-GW*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.. |
| Code Block |
Filogic-GW*CLI> pjsip show endpoints Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName..................................> Match: <criteria.........................> AorChannel: <Aor<ChannelId......................................> <State.....> <Time.....> <MaxContact> ContactExten: <Aor/ContactUri<DialedExten...............> CLCID: <ConnectedLineCID...........> <Hash....> <Status> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> > ========================================================================================== Endpoint: 1001 Match: <criteria.........................> Channel: <ChannelId......................................> <State.....> <Time.....> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================== Endpoint: 1001 Not in use 0 of inf InAuth: 1001/1001 Aor: 1001 Unavailable 0 of inf InAuth: 1001/1001 10 Aor: 1001 Contact: 1001/sip:1001@192.168.1.2:42418;transport= f6b203c8e8 NonQual 10nan Transport: udp-transport udp 0 0 0.0.0.0:5060 Endpoint: 1002 UnavailableNot in use 0 of inf InAuth: 1002/1002 Aor: 1002 10 Transport: udp-transport Contact: 1002/sip:1002@192.168.1.5:48383;transport= 0d95a96184 NonQual nan Transport: udp-transport udp 0 0 0.0.0.0:5060 Objects found: 2 Filogic-GW*CLI> |
87) Make sure that both clients and Banana PI R4 board are in the same network.
89) Add firewall rules to allow sip and rtp ports to make communication between two clients using asterisk server. Flush any rules which are blocking the clients traffic.Add rules on the top place of the chain
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root@Filogic-GW:~# iptables -F root@Filogic-GW:~# iptables -A I INPUT -p udp --dport 5060 -j ACCEPT root@Filogic-GW:~# iptables -AI INPUT -p udp --dport 10000:20000 -j ACCEPT root@Filogic-GW:~# iptables -nL Chain INPUT (policy DROP) target prot opt source destination ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpts:10000:20000 |
910) For any debug regarding the logs please set pjsip logger to on.
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pjsip set logger on |
1011) Now initiate a call in between two clients and verify the logs in asterisk CLI. Voice call will be generated and two way communication in between two clients can be happened successfully.
...
6) Add dialplan configuration in extensions.conf to route outgoing calls through this PJSIP trunk. Assuming users to dial '9' followed by the number to call externally.
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[internalfrom-usersexternal] exten => _9X.2931,1,NoOp exten => _9X.,n,Dial(PJSIP/${EXTEN:1}@linphone_freesip_trunk) exten => _9X.,n,Hangup() include => outbound-linphone Dial(PJSIP/1001&PJSIP/1002) [from-internal] exten => 1001,1,Dial(PJSIP/1001,20) exten => 1002,1,Dial(PJSIP/1002,20) include => outbound-linphone [outbound-linphone] exten => _9XX.,1,NoOp same => n,Dial(PJSIP/${EXTEN:1}@linphone_freesip_trunk,60) same => n,Hangup() |
7) Add dialplan configuration for Incoming calls from public Internet.
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[from-linphone]
exten => s,1,NoOp
same => n,Dial(PJSIP/1001,20)
same => n,Hangup()
exten => 2931,1,NoOp
same => n,Dial(PJSIP/1001,20)
same => n,Hangup() |
) |
7) After updating the configurations, reload 8) After updating the configurations, reload both the configurations from asterisk CLI.
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Filogic-GW*CLI> module reload res_pjsip.so
Module 'res_pjsip.so' reloaded successfully.
-- Reloading module 'res_pjsip.so' (Basic SIP resource)
Filogic-GW*CLI> dialplan reload
Dialplan reloaded.
[Jul 17 12:20:40] WARNING[1027809]: pbx.c:7126 add_priority: Extension '1001' priority 7 in 'default', label 'unavail' already in use at priority 6
[Jul 17 12:20:40] WARNING[1027809]: pbx.c:7126 add_priority: Extension '1002' priority 5 in 'default', label 'busy' already in use at priority 4
[Jul 17 12:20:40] WARNING[1027809]: pbx.c:7126 add_priority: Extension '1002' priority 7 in 'default', label 'unavail' already in use at priority 6
-- Time to scan old dialplan and merge leftovers back into the new: 0.000062 sec
-- Time to restore hints and swap in new dialplan: 0.000018 sec
-- Time to delete the old dialplan: 0.000046 sec
-- Total time merge_contexts_delete: 0.000126 sec
-- pbx_config successfully loaded 6 contexts (enable debug for details). |
98) Verify that the endpoints are registered successfully or not.
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Filogic-GW*CLI> pjsip show registrations
<Registration/ServerURI..............................> <Auth....................> <Status.......>
==========================================================================================
linphone_freesip_registration/sip:sip.linphone.org linphone_freesip_auth Registered (exp. 68882s ago)
Objects found: 1
Filogic-GW*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <criteria.........................>
Channel: <ChannelId......................................> <State.....> <Time.....>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
==========================================================================================
Endpoint: 1001 Not in use 0 of inf
InAuth: 1001/1001
Aor: 1001 10
Transport: udp-transport udp 0 0 0.0.0.0:5060
Endpoint: 1002 Unavailable 0 of inf
InAuth: 1002/1002
Aor: 1002 10
Transport: udp-transport udp 0 0 0.0.0.0:5060
Endpoint: linphone_freesip_trunk Not in use 0 of inf
OutAuth: linphone_freesip_auth/2931
InAuth: linphone_freesip_auth/2931
Aor: linphone_freesip_aor 1
Contact: linphone_freesip_aor/sip:sip.linphone.org: 8248a8908d Avail 187.774
Transport: udp-transport udp 0 0 0.0.0.0:5060
Identify: linphone_identify/linphone_freesip_trunk
Match: 5.135.215.43/32
Objects found: 3
Filogic-GW*CLI> |
109) Add firewall rules to allow sip and rtp ports. Flush any rules which are blocking the clients traffic.
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| Code Block |
root@Filogic-GW:~# iptables -F root@Filogic-GW:~# iptables -A I INPUT -p udp --dport 5060 -j ACCEPT root@Filogic-GW:~# iptables -AI INPUT -p udp --dport 10000:20000 -j ACCEPT root@Filogic-GW:~# iptables -nL Chain INPUT (policy DROP) target prot opt source destination ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpts:10000:20000 |
1110) Now initiate an outbound call from internal extension to the external sip server(2931).
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-- Executing [92931@from-internal:1] NoOp("PJSIP/1001-00000001", "") in new stack
-- Executing [92931@from-internal:2] Dial("PJSIP/1001-00000001", "PJSIP/2931@linphone_freesip_trunk,60") in new stack
-- Called PJSIP/2931@linphone_freesip_trunk
-- PJSIP/linphone_freesip_trunk-00000002 is ringing
-- Added contact 'sip:1001@192.168.1.9:36312;transport=UDP;rinstance=faa99161e70850c2' to AOR '1001' with expiration of 60 seconds
[Jul 16 11:20:45] WARNING[1830]: db.c:316 db_execute_sql: Error executing SQL (COMMIT): database is locked
> 0x7f940598d0 -- Strict RTP learning after remote address set to: 5.135.215.43:18034
-- PJSIP/linphone_freesip_trunk-00000002 answered PJSIP/1001-00000001
> 0x7f94049790 -- Strict RTP learning after remote address set to: 192.168.1.9:36687
-- Channel PJSIP/linphone_freesip_trunk-00000002 joined 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
-- Channel PJSIP/1001-00000001 joined 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
-- Channel PJSIP/1001-00000001 joined 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
-- Channel PJSIP/linphone_freesip_trunk-00000002 left 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
-- Channel PJSIP/1001-00000001 left 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
== Spawn extension (from-internal, 92931, 2) exited non-zero on 'PJSIP/1001-00000001'
-- Added contact 'sip:1001@192.168.1.9:36312;transport=UDP;rinstance=faa99161e70850c2' to AOR '1001' with expiration of 60 seconds basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
-- Channel PJSIP/linphone_freesip_trunk-00000002 left 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
-- Channel PJSIP/1001-00000001 left 'simple_bridge' basic-bridge <97c41ec6-e3f4-45e8-ad94-f2f8763abff7>
== Spawn extension (from-internal, 92931, 2) exited non-zero on 'PJSIP/1001-00000001'
-- Added contact 'sip:1001@192.168.1.9:36312;transport=UDP;rinstance=faa99161e70850c2' to AOR '1001' with expiration of 60 seconds |
SIP registration commands after adding the code changes:
1) By default inbound call configuration will be updated in the device.
2) Please take two clients install an application which supports voice calls and register those clients with the configuration in the "/etc/asterisk" folder.
As per current configuration,
Client1:
Host: <erouter IP>
Username: 601
Password: 601
Client2:
Host: <erouter IP>
Usename: 602
Password: 602
3) Please set below commands from the console to update the outbound configuration.
dmcli eRT setv Device.Services.VoiceService.1.VoiceProfile.1.Line.1.Enable string Enabled
dmcli eRT setv Device.Services.VoiceService.1.VoiceProfile.1.SIP.OutboundProxy string sip.linphone.org
dmcli eRT setv Device.Services.VoiceService.1.VoiceProfile.1.SIP.OutboundProxyPort uint 5060
dmcli eRT setv Device.Services.VoiceService.1.VoiceProfile.1.Line.1.SIP.AuthUserName string <username>
dmcli eRT setv Device.Services.VoiceService.1.VoiceProfile.1.Line.1.SIP.AuthPassword string <password>
dmcli eRT getv Device.Services.VoiceService.1.VoiceProfile.1.Line.1.Status
4) Please take some other client and create an account in sip.linphone.org
5) Provide the username and password in the data models as per the data used for account creation using linphone.
6) Added below logs for the reference.
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root@Filogic-GW:~# asterisk -x "pjsip show endpoints"
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <criteria.........................>
Channel: <ChannelId......................................> <State.....> <Time.....>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
==========================================================================================
Endpoint: 2931_trunk Not in use 0 of inf
OutAuth: 2931_auth/2931
Aor: 2931_aor 0
Contact: 2931_aor/sip:sip.linphone.org e53c3ca51a NonQual nan
Transport: udp-transport udp 0 0 0.0.0.0:5060
Identify: 2931_identify/2931_trunk
Match: 5.135.215.43/32
Match: 2001:41d0:203:7fe2::4/128
Match: 176.31.149.179/32
Match: 2001:41d0:303:b0d2::4/128
Endpoint: 601 Not in use 0 of inf
InAuth: 601/601
Aor: 601 10
Contact: 601/sip:601@192.168.1.2:38116;transport=UD ba3b650e9b NonQual nan
Transport: udp-transport udp 0 0 0.0.0.0:5060
Endpoint: 602 Not in use 0 of inf
InAuth: 602/602
Aor: 602 10
Contact: 602/sip:602@192.168.1.5:57429;transport=UD 2936e37328 NonQual nan
Transport: udp-transport udp 0 0 0.0.0.0:5060
Objects found: 3
root@Filogic-GW:~#
root@Filogic-GW:~#
root@Filogic-GW:~# asterisk -x "pjsip show registrations"
<Registration/ServerURI..............................> <Auth....................> <Status.......>
==========================================================================================
2931_registration/sip:sip.linphone.org 2931_auth Registered (exp. 79s)
Objects found: 1
root@Filogic-GW:~# |
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