ast: spacing & trunk defaults

asterisk/conf/mobile/pjsip.conf

@@ -11,7 +11,7 @@ type = global
 endpoint_identifier_order = ip,username,anonymous
 
 ; Basic UDP Transport.
-[ipv4-udp-nat]
+[ipv4-udp]
 type = transport
 bind = 0.0.0.0:65534
 protocol = udp
@@ -19,8 +19,3 @@ local_net = 172.16.0.0/12
 local_net = 192.168.0.0/16
 local_net = 127.0.0.0/8
 local_net = 10.0.0.0/8
-
-;
-; Load trunk identify block.
-#tryinclude pjsip_identify_trunk_one.conf
-

asterisk/conf/mobile/pjsip_users.conf

@@ -2,11 +2,35 @@
 ; /etc/asterisk/pjsip_users.conf
 ;
 [1000](extension-defaults)
+; Dialplan context name for calls originating from this account.
+endpoint/context = from-ext
 
+; Voicemail address.
+endpoint/mailboxes = 1000@default
+
+; Internal Caller ID string for this device.
+endpoint/callerid = 1000 <1000>
+
+; Username for SIP account. By convention, this should be the extension number.
 inbound_auth/username = 1000
 
+; Password for SIP account (you can choose whatever password you like).
 inbound_auth/password = MapleCarrotBlueGrainFishSoapSoup
 
-endpoint/callerid=1000 <Ext 1000>
+; Maximum number of simultaneous logins for this account.
+aor/max_contacts = 1
 
-endpoint/context=from-ext
+; Check connectivity every 30 seconds.
+aor/qualify_frequency = 30
+
+; Set connectivity check timeout to 3 seconds.
+aor/qualify_timeout = 3.0
+
+; IMPORTANT! This setting determines whether the audio stream will be proxied
+; through the Asterisk server.
+;
+; If this device is directly reachable by the internet (either by a publicly
+; routable IP, or static port mappings on your router), choose YES.
+;
+; Otherwise, if this device is hidden behind NAT, choose NO.
+endpoint/direct_media = no

asterisk/conf/mobile/pjsip_wizard.conf

@@ -1,11 +1,35 @@
 ;
 ; /etc/asterisk/pjsip_wizard.conf
+;
+[trunk-defaults](!)
+type = wizard
+
+; Send media to the address and port on the incoming packet, regardless of what
+; the SIP headers say (NAT workaround).
+endpoint/rtp_symmetric = yes
+
+; Rewrite the SIP contact to the address and port of the request (NAT workaround).
+endpoint/rewrite_contact = yes
+
+; Send the Remote-Party-ID SIP header. Some providers need this.
+endpoint/send_rpid = yes
+
+; We use alaw mainly in Europe.
+endpoint/allow = !all,alaw
+
+; Call encryption.
+endpoint/media_encryption = no
+
+; Default language, pjsip wizard appears to ignore global setting.
+endpoint/language = en_GB
+
 ;
 [extension-defaults](!)
 type = wizard
 
 ; Users must register and auth!
 accepts_registrations = yes
+
 accepts_auth = yes
 
 ; Check ext connectivity every x seconds.

asterisk/conf/mobile/quectel.conf

@@ -83,12 +83,3 @@ exten=+447123456789
 
 audio = /dev/ttyUSB1		; tty port for audio connection; 	no default value
 data = /dev/ttyUSB2		; tty port for AT commands; 		no default value
-
-; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345
-;  imei and imsi must contain exactly 15 digits !
-;  imei/imsi discovery is available on Linux only
-;imei=123456789012345
-;imsi=123456789012345
-
-; if audio and data set together with imei and/or imsi audio and data has precedence
-;   you can use both imei and imsi together in this case exact match by imei and imsi required