[asterisk-commits] oej: branch oej/pgtips-srv-and-outbound-stuff-1.8 r386003 - /team/oej/pgtips-...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Apr 18 09:35:49 CDT 2013


Author: oej
Date: Thu Apr 18 09:35:48 2013
New Revision: 386003

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=386003
Log:
Adding a question that I want you to answer. Thank you for your feedback.

Modified:
    team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records

Modified: team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records
URL: http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records?view=diff&rev=386003&r1=386002&r2=386003
==============================================================================
--- team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records (original)
+++ team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records Thu Apr 18 09:35:48 2013
@@ -38,4 +38,32 @@
 This is not in scope for this project. Work is in progress in the SIP Forum IPv6 wg
 to define a process for this situation.
 
-This work is funded by Inteno, Sweden.
+This work is funded by Inteno Broadband Solutions AB, Sweden. http://www.intenogroup.com
+
+Asterisk changes (brainstorm suggestion)
+========================================
+
+- Make sure that the SIP channel gets a list on a SRV lookup, not a single host.
+- Add failover based on that list
+- Make sure the list is searched on every transaction and have failover in a call
+
+Peer matching
+-------------
+This is a bit more interesting. If you connect to a service provider and add
+this to sip.conf
+
+register=user:password at serviceprovider/1234
+
+
+[serviceprovider]
+type=peer
+host=serviceproviderdomain.tld
+context=bazinga
+
+Yyou expect to be able to receive calls from the service provider
+in that peer. Now the service provider has a list of servers using IPv4
+and IPv6 in the SRV records for that name. Should the Asterisk SIP channel
+match the full list?
+Currently Asterisk will resolve to one IP address and use that for matching
+of incoming calls, possibly deny calls from servers that does not match.
+This is NOT the expected solution.




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