path: root/funcs/func_srv.c
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2011-07-14Merged revisions 328247 via svnmerge from lmadsen1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Commit compromise I suggested on review 608.mmichelson1-0/+14
This allows for multiple SRV queries to be done from the dialplan for the same service on a single call while still allowing one to bypass the call to SRVQUERY if they so please. Taking action since no comments had been left for a while. This can easily be reverted if needed. External tests still pass. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257851 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-13Address Russell's comments on func_srv from reviewboard.mmichelson1-7/+14
* Change copyright date * Place channel in autoservice when doing SRV lookup * Get rid of trailing whitespace * Change logic in load_module function git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257025 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Fix some compiler errors that popped up after the CCSS merge.mmichelson1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256529 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09func_srv and explicit specification of a remote IP for SIP.mmichelson1-0/+254
From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256485 f38db490-d61c-443f-a65b-d21fe96a405b