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SipDevicesAsterisk » History » Revision 1

Revision 1/4 | Next »
Anonymous, 11/14/2010 11:22 AM


Asterisk PBX

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.

=== Versions 1.4 and 1.6.x ===

'''dnsmgr.conf'''

{{{
[general]
enable=yes
}}}

'''sip.conf'''

{{{
[general]
...
srvlookup=yes
...

register => 2233XXXXX:/2233XXXXX
...

[authentication]

[sip2sip](!)
type=peer
canreinvite=no
nat=yes
qualify=yes
domain=sip2sip.info
fromdomain=sip2sip.info
outboundproxy=proxy.sipthor.net
fromuser=2233XXXXX
username=2233XXXXX
secret=password
insecure=invite
context=from-sip2sip

[sip2sip-0](sip2sip)
host=sip2sip.info

[sip2sip-1](sip2sip)
host=81.23.228.129

[sip2sip-2](sip2sip)
host=81.23.228.150

[sip2sip-3](sip2sip)
host=85.17.186.7
}}}

'''extensions.conf''' {{{
[from-users]
; Dialing the SIP2SIP echo test
; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer
exten => 1234,1,Dial(SIP/3333@sip2sip-0)

[from-sip2sip]
; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern
exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID})
exten => 2233XXXXX,n,Dial(SIP/phone1, 60)
}}}

=== Version 1.8 ===

'''dnsmgr.conf'''

{{{
[general]
enable=yes
}}}

'''sip.conf'''

{{{
[general]
...
srvlookup=yes
...

register => 2233XXXXX:/2233XXXXX
...

[authentication]

[sip2sip](!)
type=peer
canreinvite=no
nat=yes
qualify=yes
domain=sip2sip.info
fromdomain=sip2sip.info
outboundproxy=proxy.sipthor.net
fromuser=2233XXXXX
defaultuser=2233XXXXX
secret=password
insecure=invite
context=from-sip2sip

[sip2sip-0](sip2sip)
host=sip2sip.info

[sip2sip-1](sip2sip)
host=81.23.228.129

[sip2sip-2](sip2sip)
host=81.23.228.150

[sip2sip-3](sip2sip)
host=85.17.186.7
}}}

'''extensions.conf''' {{{
[from-users]
; Dialing the SIP2SIP echo test
; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer
exten => 1234,1,Dial(SIP/3333@sip2sip-0)

[from-sip2sip]
; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern
exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID})
same => n,Dial(SIP/phone1, 60)
}}}

Updated by about 14 years ago · 1 revisions