SipDevicesAsterisk

Version 1 (Anonymous, 11/14/2010 11:22 am)

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== Asterisk PBX ==
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The SIP2SIP platform consists of several servers, addressed by DNS SRV records. Asterisk, however is currently unable to handle more that one result for a DNS SRV lookup, so the configuration needed for getting it work with the SIP2SIP service is a bit confusing. This wiki page helps clarify that hopefully.
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=== Versions 1.4 and 1.6.x ===
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'''dnsmgr.conf'''
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{{{
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[general]
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enable=yes
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}}}
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'''sip.conf'''
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{{{
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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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}}}
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'''extensions.conf'''
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{{{
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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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}}}
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=== Version 1.8 ===
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'''dnsmgr.conf'''
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{{{
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[general]
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enable=yes
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}}}
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'''sip.conf'''
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{{{
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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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}}}
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'''extensions.conf'''
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{{{
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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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}}}