Commit Graph

3370 Commits

Author SHA1 Message Date
Joshua Colp
c126127fd5 Merged revisions 117574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines

Apply the autoframing setting to dialogs that do not get matched against a user or peer.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-21 19:39:42 +00:00
Russell Bryant
29a9d477df Remove duplicate colon on Reason header
(closes issue #12678)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-18 19:58:10 +00:00
Joshua Colp
30aedbade7 Try to fix attended transfers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-16 21:34:45 +00:00
Jeff Peeler
f97d547aba Fixes a problem I was having with two SIP phones using Packet2Packet bridging dropping audio nearly immediately. The problem was that the lock on the SIP dialog was not being unlocked while the bridge was still active. (Related to issue #12566)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-15 21:54:18 +00:00
Joshua Colp
46423f6e09 Fix pedanticness.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 21:54:03 +00:00
Olle Johansson
eecea3268e Don't add linefeed on received MESSAGE
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 14:16:51 +00:00
Olle Johansson
f07454f25d Properly declare charset for text messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 14:03:42 +00:00
Olle Johansson
bb386c84e7 Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 13:37:07 +00:00
Olle Johansson
47bf217ee8 Merged revisions 116230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines

Accept text messages even with
Content-Type: text/plain;charset=Södermanländska

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 13:05:15 +00:00
Olle Johansson
29b1d73567 Add support for codec settings in originate via call file and manager.
This is to enable video and text in originated calls. Development sponsored
by Omnitor AB, Sweden. (http://www.omnitor.se)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 12:32:57 +00:00
Olle Johansson
9c2956a3b0 Reformatting
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 11:37:21 +00:00
Olle Johansson
615ed013d3 Adding comments
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14 11:32:05 +00:00
Mark Michelson
7daebcd610 Adding support for "urgent" voicemail messages. Messages which are
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.

There are two ways to leave an urgent message. 
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for 
   a caller to mark a message as urgent after the message has been recorded.

I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.

(closes issue #11817)
Reported by: jaroth
	Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-09 21:22:42 +00:00
Russell Bryant
c02cf176e1 Merged revisions 115561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines

Don't give up on attempting an outbound registration if we receive a 408 Timeout.
(closes issue #12323)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-08 16:14:08 +00:00
Joshua Colp
4555f32184 Remove redundant header getting.
(closes issue #12597)
Reported by: hooi


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-07 13:41:25 +00:00
Russell Bryant
e9f62e1d41 Change some NOTICE log messages to debug.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-06 15:14:55 +00:00
Russell Bryant
2a966cdb03 Merged revisions 115304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines

Avoid putting opaque="" in Digest authentication.  This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-05 19:50:24 +00:00
Tilghman Lesher
b11854445b Add attributes to various API calls, to help track down bugs (and remove a deprecated function)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-02 02:33:04 +00:00
Joshua Colp
f4237076bf Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30 20:51:17 +00:00
Olle Johansson
4c3aecfc55 Merged revisions 114890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines

Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)

(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30 16:55:49 +00:00
Tilghman Lesher
72b5d8d982 Unleak reference
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-26 15:08:51 +00:00
Tilghman Lesher
c5f11a59d0 Add 'sip qualify peer <peer>' command (with AMI SIPqualifypeer)
(closes issue #12524)
 Reported by: ctooley
 Patches: 
       sip_qualify_peer.diff.2 uploaded by ctooley (license 136)
       some modifications for trunk by Corydon76
 Tested by: Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-26 02:48:56 +00:00
Michiel van Baak
08e674bce0 Pass the hangup cause all the way to the calling app/channel.
(closes issue #11328)
Reported by: rain
Patches:
      20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 22:16:48 +00:00
Joshua Colp
a50b48dacd Hey look, it builds.
(closes issue #12519)
Reported by: falves11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 22:11:46 +00:00
Mark Michelson
cb80defb68 Merged revisions 114632 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines

Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.

(closes issue #12513)
Reported by: mneuhauser
Patches:
      asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 21:35:39 +00:00
Olle Johansson
9a4e9f5944 Merged revisions 114603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines

Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 14:59:05 +00:00
Russell Bryant
767fa7a909 Change a verbose message to debug.
(closes issue #12514)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 14:55:21 +00:00
Olle Johansson
2958831a97 Merged revisions 114584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines

Add 502 support for both directions, not only one...  (see r114571)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-23 16:53:34 +00:00
Tilghman Lesher
b170c36350 Merged revisions 114571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines

Treat a 502 just like a 503, when it comes to processing a response code

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 23:58:19 +00:00
Joshua Colp
1e066813ac Add support for authenticating on a NOTIFY request. This is useful for phones that require it when sending them a special packet to get them to do something (such as reload their configuration).
(closes issue #9896)
Reported by: IgorG
Patches:
      sipnotify-113980-v14.patch uploaded by IgorG (license 20)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 15:54:06 +00:00
Steve Murphy
161b4abd79 Hopefully, this will resolve the issues that russellb had with this log_show_lock().
I gathered the code that filled the string, and put it in a different func which
I cryptically call "append_lock_information()".
Now, both log_show_lock(), and handle_show_locks() both call this code to do
the work. Tested, seems to work fine. 
Also, log_show_lock was modified to use the ast_str stuff, along with checking
for successful ast_str creation, and freeing the ast_str obj when finished.
A break was inserted to terminate the search for the lock; we should never
see it twice.

An example usage in chan_sip.c was created as a comment, for instructional
purposes.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 14:38:46 +00:00
Jeff Peeler
41fd7a6a21 (closes issue #6113)
Reported by: oej
Tested by: jpeeler

This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.

Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 23:42:45 +00:00
Joshua Colp
a79214b5b1 Merged revisions 114322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines

Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 14:40:33 +00:00
Sean Bright
e4dce85331 Merged revisions 114245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line

Only complete the SIP channel name once for 'sip show channel <channel>'
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18 13:38:07 +00:00
Steve Murphy
5203c664de Thanks to snuff for finding these omissions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-17 14:45:16 +00:00
Steve Murphy
5fb4b1bbe5 This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16 23:53:27 +00:00
Olle Johansson
18866623dc Merged revisions 114148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines

Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15 20:39:29 +00:00
Olle Johansson
f239f24580 Adding chanvar to SIPPEER from 1.4 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15 20:31:08 +00:00
Joshua Colp
c5d0ca23f0 Merged revisions 114103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines

It is possible for the remote side to say they want T38 but not give any capabilities.
(closes issue #12414)
Reported by: MVF

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14 14:53:33 +00:00
Mark Michelson
d13b45564b Merged revisions 114045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines

Be sure that we're not about to set bridgepvt NULL prior to dereferencing it.

(closes issue #11775)
Reported by: fujin


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 19:58:36 +00:00
Joshua Colp
a4e73acaf8 Merged revisions 114021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines

Don't add custom URI options if they don't exist OR they are empty.
(closes issue #12407)
Reported by: homesick
Patches:
      uri_options-1.4.diff uploaded by homesick (license 91)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 13:28:30 +00:00
Mark Michelson
88cc98ea94 Merged revisions 113927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines

We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.

(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 20:56:14 +00:00
Mark Michelson
925924386a Merged revisions 113681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines

If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.

(closes issue #12392)
Reported by: fnordian
Patches:
      chan_sip.patch uploaded by fnordian (license 110) with small modification from me


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 14:41:58 +00:00
Tilghman Lesher
fa875c0578 Merged revisions 113348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines

Move check for still-bridged channels out a little further, to avoid possible
deadlocks.  (Closes issue #12252)
Reported by: callguy
 Patches: 
       20080319__bug12252.diff.txt uploaded by Corydon76 (license 14)
 Tested by: callguy

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08 15:48:58 +00:00
Jeff Peeler
bb13bf705e Merged revisions 113013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines

Merged revisions 113012 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines

(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 21:35:48 +00:00
Jeff Peeler
566e073606 Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines

(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 15:18:10 +00:00
Steve Murphy
f291c2af0a Found a little problem with the sip request handling that could lead to a quick crash of asterisk, and a road to a DOS attack if left unfixed.
Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes.

I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short. 
Instead of freeing the struct and nulling the pointer, it now just resets it, because this 
ast_str is expected by the calling routine to still be there after handle_request_do() returns.

This appears to fix the crash. I assume that it was introduced with ast_str's being adopted.  It's a subtle and easy-to-miss sort of problem.

I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well; 
no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do...




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-05 01:33:13 +00:00
Joshua Colp
b5cccfe1a4 Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off.
(closes issue #12169)
Reported by: pj
Patches:
      12169-debugoff-2.diff uploaded by qwell (license 4)
Tested by: pj


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02 15:26:51 +00:00
Jeff Peeler
6699761f80 Added dnsmgr status output for sip show registry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 22:55:28 +00:00
Jeff Peeler
a5cdd849e5 This adds DNS SRV record support to DNS manager. If there is a SRV record for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 17:53:08 +00:00