Asterisk PBX
Asterisk PBX
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.
Versions 1.4 and 1.6.x
dnsmgr.conf
[general] enable=yes
sip.conf
[general] ... srvlookup=yes ... register => 2233XXXXX:password@sip2sip.info/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(all)})
exten => 2233XXXXX,n,Dial(SIP/phone1, 60)Version 1.8
dnsmgr.conf
[general] enable=yes
sip.conf
[general] ... srvlookup=yes ... register => 2233XXXXX:password@sip2sip.info/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(all)})
same => n,Dial(SIP/phone1, 60)Tags
None
Referenced Files
None
Subscribers
None
- Last Author
- tijmen
- Last Edited
- Dec 27 2016, 12:16 PM