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Next, we have to setup SIP side of our gateway. We will do this in /etc/asterisk/sip.conf.
We will not allow unauthenticated clients (we will route them into context guest, see above).
. Set your realm and domain to something usefull which should be unique. Putting your real domain there is realy good point of start. But remember, this domain and realm setting has to corespond to settings in client, see above. Only opensource codecs are enabled by default. If you buy g.729 codec, you can enable it. Again, set language to fit your needs.
Localnets are defined to better recognisation of NAT. Lines starting with jb are turning on jitterbuffer which is good to enable. Next, we define phones which we will register. In example, we use 3-digit dialplan and we are starting with number 200. Define next sip clients to fit your needs. It is good if username corespond with peer name (name in []).
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[general] context=guest ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls maxexpirey=360 ; Max length of incoming registration we allow defaultexpirey=300 ; Default length of incoming/outoing registration t38udptlsupport=yes ; Turn on support for T.38 UDPTLrealm=somecompany.com domain=somecompany.com disallow=all ; First disallow all codecs ;allow=g729 allow=gsm allow=alaw allow=ulaw language=xx ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet=169.254.0.0/255.255.0.0 ;Zero conf local network jbenable=yes jbforce=yes jbimpl = fixed [200] username=200 secret=somesecret200 type=friend host=dynamic context=sip [...] ... ... |