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SipDevicesAsterisk » History » Version 4

Adrian Georgescu, 10/15/2011 05:15 PM

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h2. Asterisk PBX
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SIP2SIP service consists of several OpenSIPS servers running in difference data centers using SIP Thor distributed architecture, which is using DNS SRV records to point to the actual servers. Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial. This wiki page helps clarify it.
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h3. Versions 1.4 and 1.6.x
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*dnsmgr.conf*
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<pre>
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[general]
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enable=yes
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</pre>
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*sip.conf*
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<pre>
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[general]
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...
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srvlookup=yes
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...
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register => 2233XXXXX:password@sip2sip.info/2233XXXXX
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...
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[authentication]
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[sip2sip](!)
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type=peer
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canreinvite=no
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nat=yes
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qualify=yes
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domain=sip2sip.info
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fromdomain=sip2sip.info
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outboundproxy=proxy.sipthor.net
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fromuser=2233XXXXX
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username=2233XXXXX
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secret=password
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insecure=invite
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context=from-sip2sip
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[sip2sip-0](sip2sip)
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host=sip2sip.info
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[sip2sip-1](sip2sip)
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host=81.23.228.129
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[sip2sip-2](sip2sip)
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host=81.23.228.150
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[sip2sip-3](sip2sip)
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host=85.17.186.7
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</pre>
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*extensions.conf*
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<pre>
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[from-users]
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; Dialing the SIP2SIP echo test
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; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer
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exten => 1234,1,Dial(SIP/3333@sip2sip-0)
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[from-sip2sip]
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; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern
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exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID(all)})
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exten => 2233XXXXX,n,Dial(SIP/phone1, 60)
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</pre>
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h3. Version 1.8
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*dnsmgr.conf*
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<pre>
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[general]
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enable=yes
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</pre>
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*sip.conf*
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<pre>
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[general]
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...
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srvlookup=yes
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...
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register => 2233XXXXX:password@sip2sip.info/2233XXXXX
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...
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[authentication]
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[sip2sip](!)
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type=peer
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canreinvite=no
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nat=yes
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qualify=yes
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domain=sip2sip.info
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fromdomain=sip2sip.info
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outboundproxy=proxy.sipthor.net
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fromuser=2233XXXXX
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defaultuser=2233XXXXX
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secret=password
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insecure=invite
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context=from-sip2sip
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[sip2sip-0](sip2sip)
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host=sip2sip.info
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[sip2sip-1](sip2sip)
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host=81.23.228.129
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[sip2sip-2](sip2sip)
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host=81.23.228.150
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[sip2sip-3](sip2sip)
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host=85.17.186.7
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</pre>
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*extensions.conf*
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<pre>
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[from-users]
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; Dialing the SIP2SIP echo test
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; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer
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exten => 1234,1,Dial(SIP/3333@sip2sip-0)
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[from-sip2sip]
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; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern
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exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID(all)})
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same => n,Dial(SIP/phone1, 60)
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</pre>