sip.conf file configuration

In the sip.conf file we can configure everything related with the SIP protocol; add new sip users or define sip providers.

For example, and easy example of the sip.conf file:

[general]
context=default
port=5060 ; UDP port for Asterisk
bindaddr=0.0.0.0 ; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP
srvlookup=yes ; Enable DNS SRV server

[peter]
type=friend
secret=welcome
qualify=yes ;latency must be under 2000ms.
nat=no ; Telephone without NAT
host=dynamic ; the devices can be registered with different IPs each time
canreinvite=no ; Asterisk by default redirects
context=internal ; the context of the extensions.conf file

The sip.conf file starts with a [general] section with the default configuration for every user and peer (providers). The default values can be overwritten in the particular configuration of each user or peer

- In general, SIP servers use port 5060 UDP. That is why we use port=5060 . Sometimes, for example if we use SER (Sip Express Router) with Asterisk we should change the port number.

- DNS is a way to manage a logic adress in order to be resolve. That allows, that calls can be led to different places without changing the logic address. Using DNS SRV we have the advantages of DNS whereas if we give the "no" value it is not possible to route calls based in domains. It is a good idea to active the service with srvlookup=yes

Each extension is defined by a user or a peer or a friend and is called with a name between []

- "user" type is used to authenticate incoming calls, "peer" for outcoming calls and "friend" for both. In the example we have a "friend" extension called "peter". It can make incoming and outcoming calls.

- "secret" is the password used to authenticate. In this case it will be "welcome"

- The latency between Asterisk server and the telephone can be monitored with qualify=yes to determine when the device can be reached. Asterisk considers by default that a device is present if its latency is smaller than 2000 ms (2 seconds). This value can be changed with the number of milliseconds instead of "yes".

- If an extension is behind a device that makes NAT (Network Address Translation) like a router or a firewall "nat=yes" force Asterisk to ignore the field contact information and it will use the address which the packages come from.

- If we put "host=dynamic" means that the telephone will be able to connect from any IP address. We can limit this user to access with only an IP or a domain name. If we put "host=static" it would not be necessary that the user will register itself with the password provided in "secret".

- In SIP, invite messages are used to establish calls and to redirect audio or video. Any invite message after initial invite message in the same conversation is considered a reinvite. For example, two users are connected and one of them active Music on Hold (MoH) because he wants to stop the conversation. Therefore, Asterisk make a reinvite to the second user. Later, the first user wants to follow the call and send a reinvite message to Asterisk and is sent to the second user and both are again connected. Using canreinvite=no. force Asterisk to be in the middle not allowing that the final points interchange messages RTP directly.

Finally context=internal shows the context where the instructions for this extension will be executed. This is related to the context of the file extensions.conf that gives the dialplan for that context. Therefore the "internal" context must exist in the file extensions.conf or otherwise we should not create it. Several extensions can have the same context.

Advanced Options:

In the following columns we have the possibilities to configure the "user" and "peer" types . In the case of "friend" both columns are possible because a friend is a user + a peer.

Ejemplos:

[grandstream1]
type=friend ; is both peer and user
context=mycontext ; context name
username=grandstream1 ; usually the same as the section
fromuser=grandstream1 ; overwrites caller-ID
callerid=John Smith<1234>
host=192.168.0.23 ; a private IP of a LAN
nat=no ; no NAT
canreinvite=yes ;
dtmfmode=info ; it can be RFC2833 or INFO
mailbox=1234@default ; mailbox 1234 in the context "default"
disallow=all ;
allow=ulaw ; allow alaw codec
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Only g723.1 pass through
;allow=g729 ; Only g729 pass through

[xlite1]
;silence suppresion can be activated
;Xlite sends keep-alive NAT packets ,so qualify=yes is not necessary
type=friend
username=xlite1
callerid="john jones" <5678>
host=dynamic ; xlite softphone could be in any IP address
nat=yes ; X-Lite is behind a NAT device
canreinvite=no ; when a device is behind a NAT device it usually is no
disallow=all
allow=gsm ; GSM needs low bandwithd than ulaw and alaw
allow=ulaw
allow=alaw

[user1_snomsip]
type=friend
secret=blah ; password to register
host=dynamic
dtmfmode=inband ; the possibilities are inband , rfc2833, or info
defaultip=192.168.0.59 ; IP address of the device
mailbox=1234; Voicemail
disallow=all
allow=ulaw ; because we have chosen inband for dtmf we need alaw or ulaw (G.711)
allow=alaw

[user2_pingtel]
type=friend
username=user2_pingtel
secret=blah
host=dynamic
qualify=1000 ; If it is over 1 second without response the connection is broken
callgroup=1,3-4 ; members of groups 1,3 and 4
pickupgroup=1,3-4 ; member of "pickup" groups 1,2 and 4
defaultip=192.168.0.60 ;IP
disallow=all
allow=ulaw
allow=alaw
allow=g729

[user3_cisco]
type=friend
username=user3_cisco
secret=blah
nat=yes ; is behind NAT
host=dynamic
canreinvite=no ;
qualify=200 ; 200 ms to receive a response
defaultip=192.168.0.4
disallow=all
allow=ulaw
allow=alaw
allow=g729

[user4_cisco1]
type=friendusername=user4_cisco
fromuser=peter ;
secret=blah
defaultip=192.168.0.4 ;
amaflags=default ; Possibilities are default, omit, billing or documentation
accountcode=peter ; For billing purposes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1