autoreview, editor
102
edits
VoIP Phones - Asterisk: Difference between revisions
→PJSIP: IPv6
No edit summary |
|||
| (4 intermediate revisions by the same user not shown) | |||
|
It isn't a good idea to have an installation that mixes sip.conf with pjsip.conf.
When reading the instructions below be aware which are for sip.conf and which are for pjsip.conf
=pjsip.conf (PJSIP)=
allow = ulaw
direct_media = no
rtp_symmetric = yes
aors = aaisptrunk
outbound_auth=auth_reg_442082881111
In extensions.conf you can dial out via the trunk with:
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
exten => s-BUSY,1,Playtones(busy)
exten => s-CONGESTION,1,Playtones(congestion)
exten => s-CHANUNAVAIL,1,Playtones(unobtainable)
exten => s-NOANSWER,1,Playtones(congestion)
==PJSIP: Trunk without registration==
Use the above example but do not include the top section for "[reg_442082881111]".
Then set the AAISP control panel to point to your server by hostname or IP address:<br />
[[File:Asterisk pjsip noregistration.png|border]]
==PJSIP: Keep-Alive / Anti-Idle==
contact = sip:+442082881111@voiceless.aa.net.uk
qualify_frequency=20
==PJSIP: IPv6==
By default PJSIP does not listen on IPv6.<br />
At the top of pjsip.conf you will need to add another transport to go along with your IPv4 transport (usually a section with bind set to 0.0.0.0 or your IP address):
[transport-udp6]
type=transport
protocol=udp
bind=[::]
You may need to force the endpoint to use this transport (a small section of the endpoint section above but with the "transport = " line inserted):
[aaisptrunk]
type = endpoint
transport = transport-udp6
==Status and Commands==
| |||