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Trunking.md

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Trunking

SIP account type

Use type=friend for user agents or extensions. And use type=peer for SIP proxy or gateway, or to define connection to other server.

Notes:

  • type=friend the host option must be dynamic, host=dynamic
  • type=peer the host must be defined, host=<ip address or hostname>

Example SIP account for gateway (in sort: SIP peer):

[gateway-abc]
type=peer
host=192.168.1.18
context=from-trunk
qualify=20000
disallow=all
allow=ulaw

Note the use of option context above, it is where incoming call through that peer will be handled.

In above example calls coming to SIP peer gateway-abc will be handled by dialplans in from-trunk.

Edit /etc/asterisk/sip.conf

All SIP peers will be managed from special file included by sip.conf.

Add this line to the bottom of sip.conf:

#include sip_peers.conf

Edit /etc/asterisk/sip_peers.conf

Create sip_peers.conf. You will add SIP peers in this file, instead of in sip.conf or sip_friends.conf.

vi /etc/asterisk/sip_peers.conf

Example adding 2 SIP gateways:

[gateway-sg]
type=peer
host=192.168.1.18
context=from-trunk
qualify=20000
disallow=all
allow=ulaw

[gateway-id]
type=peer
host=192.168.2.77
context=from-trunk
qualify=20000
disallow=all
allow=ulaw

Reload SIP configuration and check for peers:

asterisk -rx 'sip reload'
asterisk -rx 'sip show peers'

Asterisk Register

Asterisk can act as a SIP user agent registered to another Asterisk or other SIP server. This is very useful when Asterisk is behind a NAT router, especially when the NAT router is not configured with port forwarding for the Asterisk.

Define SIP peer

In order to get Asterisk register to other server, first we need to define the SIP peer for that server.

Example, to register this Asterisk to VoIP Rakyat:

[voiprakyat]
type=peer
host=voiprakyat.or.id
context=from-trunk
username=211222
secret=verysecret11111
qualify=yes
canreinvite=no
disallow=all
allow=gsm

Setup register

Next, add this line in sip.conf, in general block:

register = 211222:verysecret11111@voiprakyat

Basic format is:

register = <username>:<password>@<SIP peer>

Check registration status:

asterisk -rx 'sip show registry'

Detail information about this, about other formats, can be read at shipped sip.conf (now its in /etc/asterisk/sip.conf.dist). Look for OUTBOUND SIP REGISTRATIONS section.