When I first started playing around with SIP trunks and registering it. Had issues and with help, reading and playing around with the configurations. This is what I found. I decided I should share this since I have seen a few posts about configuring SIP trunks. The below configuration is only for credential based SIP trunks. The other SIP trunk is IP address based, where the calls are validated at the SIP providers end is by the Public IP.
- Please find the configurations steps below if you have multiple direct dials and would like to display each number when
the call is made. e.g. extension 2001 with Direct Dial of 01xxxxxxx2001 calls, he wants that number to be displayed.
If the below mentioned configuration is used, ensure that translation rules are in place, since the SIP provider looks
for a number that is part of the trunk to validate the call.
sip-ua
credentials username xxxxxxxxxxxxx password xxxxxxxxxxxxxx realm sip-provider.com
authentication username xxxxxxxxxxxxxxx password xxxxxxxxxxxxxx realm sip-provider.com [SI1]
no remote-party-id [SI2]
retry register 10
timers register 1000
registrar dns:sip-provider.com
expires 60
voice service voip
ip address trusted list
ipv4 <add IPs>
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip [SI3]
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 1500 min 500
no call service stop
- Say you ahve only one number in your trunk and all outgoing calls will be sent with the same number.
Under SIP-UA add the following command with the rest of the other commands above. You will not require
a translation rule for outbound since you will be sending a single number.
sip-ua
calling-info pstn-to-sip from number set 01xxxxxxxxx
- If you ahve Multiple SIP trunks dont panic, just add another credential line (similar to point 1).
I hope this document will be useful to you.
If you have any questions please do not hesitate to comment below and I will be happy to assist you.
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[SI1]This information will be provided by the SIP provider
[SI2]This is done so that the original number is stripped (sometimes in a sip to sip call the extension number is displayed)
[SI3]Allow different protocol call
