w1ve
Guru
- Joined
- Nov 15, 2007
- Messages
- 819
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- 218
I don't know much about DNS as will be evident by these questions:
Is there a possibility of conflict if I use the same URI as email address; i.e. call or email me at [email protected]?
Awesome, pls reserve [email protected] for me!
_sip._udp.yourdomain.com. IN SRV 10 10 5060 east.pbxinaflash.com.
_sip._udp.yourdomain.com. IN SRV 10 10 5060 west.pbxinaflash.com.
sip-bill.yourdomain.com. IN TXT "[email protected]"
ns1.dns-hosting.info
ns2.dns-hosting.info
ns3.dns-hosting.info
mysipdomain.com. 3600 IN NAPTR 10 100 "s" "SIPS+D2T "" _sips._tcp.mysipdomain.com.
mysipdomain.com. 3600 IN NAPTR 20 100 "s" "SIP+D2T" "" _sip._tcp.mysipdomain.com.
mysipdomain.com. 3600 IN NAPTR 30 100 "s" "SIP+D2U" "" _sip._udp.mysipdomain.com.
_sips._tcp.mysipdomain.com. 3600 IN SRV 100 100 443 proxy.sipthor.net.
_sip._tcp.mysipdomain.com. 3600 IN SRV 100 100 5060 proxy.sipthor.net.
_sip._udp.mysipdomain.com. 3600 IN SRV 100 100 5060 proxy.sipthor.net.
_msrps._tcp.mysipdomain.com. 3600 IN SRV 0 0 2855 msrprelay.sipthor.net.
_stun._udp.mysipdomain.com. 3600 IN SRV 0 0 3478 stun1.sipthor.net.
_stun._udp.mysipdomain.com. 3600 IN SRV 0 0 3478 stun2.sipthor.net.
xcap.mysipdomain.com. 3600 IN TXT "https://xcap.sipthor.net/xcap-root/"
Trunk Name: mysipdomain
PEER Details:
type=peer
secret=***your secret***
qualify=yes
outboundproxy=proxy.sipthor.net
nat=yes
insecure=invite
host=sip2sip.info&81.23.228.129&81.23.228.150&85.17.186.7
fromuser=***your sip id***
fromdomain=mysipdomain.com
domain=mysipdomain.com
defaultuser=***your sip id***
context=from-trunk
canreinvite=no
Register string:
***your sip id***:***your secret***@mysipdomain/***your sip id***
WardWhy do you want a secure SIP URI for your Asterisk server? Because all calls worldwide to SIP URIs using Asterisk, YATE, FreeSwitch or any SIP phone are free!
I have several Google Voice trunks that go through Simonics. I have found that in order to complete an inbound call, a SIP URI is necessary.
host=sip2sip.info&81.23.228.129&81.23.228.150&85.17.186.7&**mysipdomain.com**
Finally, you need to create an inbound route. Call this route mysipdomain-Incoming, and set the DID Number to ***your sip id*** from above. Point the route to your favorite extension/ivr/whatever.
Check to make sure your new trunk is registered. At this point, you should be able to receive incoming calls on your Asterisk box to your new vanity sip id from anywhere.
Dave
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