Hi, Im trying to configure a PBX with asterisk but after a lot of tryouts and configs I cant see what is the error of my configuration.
I see some error when I start asterisk
[Apr 15 07:43:58] WARNING[2726]: loader.c:548 load_dynamic_module: Error loading module ‘res_monitor.so’: /usr/lib/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop
…
[Apr 15 07:43:58] WARNING[2726]: res_phoneprov.c:1229 get_defaults: Unable to find a valid server address or name.
…
[Apr 15 07:43:58] ERROR[2726]: ari/config.c:296 process_config: No configured users for ARI
…
[Apr 15 07:43:58] WARNING[2726]: loader.c:590 load_dynamic_module: Error loading module ‘res_ari_mailboxes.so’: /usr/lib/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Apr 15 07:43:58] WARNING[2726]: loader.c:1076 load_resource: Module ‘res_ari_mailboxes.so’ could not be loaded.
Loading res_parking.so.
…
[Apr 15 07:43:58] WARNING[2726]: chan_sip.c:32060 reload_config: Cannot use ‘tls’ transport with tlsenable=no. Removing from available transports.
[Apr 15 07:43:58] ERROR[2726]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“PBX-VirtualBox”, “(null)”, …): Name or service not known
[Apr 15 07:43:58] WARNING[2726]: acl.c:800 resolve_first: Unable to lookup ‘PBX-VirtualBox’
…
== Parsing ‘/etc/asterisk/skinny.conf’: Found
[Apr 15 07:43:58] ERROR[2769]: tcptls.c:922 ast_tcptls_client_start: Unable to connect SIP socket to 127.0.0.1:40239: Connection refused
[Apr 15 07:43:58] WARNING[2726]: chan_skinny.c:8435 config_load: Unable to get our IP address, Skinny disabled
…
[Apr 15 07:43:59] WARNING[2726]: pbx_dundi.c:4844 set_config: Unable to look up host ‘PBX-VirtualBox’
I can call to a test extension and I see this lines in my log:
[Apr 15 07:44:54] NOTICE[2789]: chan_sip.c:23869 handle_response_peerpoke: Peer ‘dialer’ is now Reachable. (41ms / 2000ms)
== WebSocket connection from ‘127.0.0.1:40252’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘dialer’ at 127.0.0.1:40252
> Saved useragent “IM-client/OMA1.0 sipML5-v1.2015.03.18” for peer dialer
== Using SIP RTP CoS mark 5
– Executing [100@from-internal:1] Wait(“SIP/dialer-00000000”, “5”) in new stack
– Executing [100@from-internal:2] Answer(“SIP/dialer-00000000”, “”) in new stack
– Executing [100@from-internal:3] Wait(“SIP/dialer-00000000”, “1”) in new stack
– Executing [100@from-internal:4] Playback(“SIP/dialer-00000000”, “yeah”) in new stack
– <SIP/dialer-00000000> Playing ‘yeah.gsm’ (language ‘en’)
– Executing [100@from-internal:5] Wait(“SIP/dialer-00000000”, “1”) in new stack
– Auto fallthrough, channel ‘SIP/dialer-00000000’ status is 'UNKNOWN
This is my configuration
SIP.conf
[general]
tcpenable=yes
localnet=192.168.1.89/255.255.0.0
externip=95.39.181.125
transport=tcp,ws,tls,wss
port=5060
bindaddr=0.0.0.0
qualify=no
disable=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
srvlookup=yes
nat=force_rport,comedia
[dialer]
avpf=yes
encryption=yes
icesupport=yes
type=friend
context=from-internal
host=dynamic
secret=passwd
qualify=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
[voxbeam_outbound]
type=friend
canreinvite=no
host=sbc.voxbeam.com
[voxbeam_inbound]
type=friend
canreinvite=no
host=95.211.119.240
can anoyone help me?
Thanks!