Age | Commit message (Collapse) | Author |
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(closes issue #13346)
Reported by: srt
Patches:
13346_malformed_sip_show_peer_response.diff uploaded by srt (license 378)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines
sip_read should properly handle a NULL return from sip_rtp_read.
(closes issue #13257)
Reported by: travishein
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breaking existing frameworks). Pointed out by Laureano on #asterisk-dev.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines
More fixes for realtime peers.
(closes issue #12921)
Reported by: Nuitari
Patches:
20080804__bug12921.diff.txt uploaded by Corydon76 (license 14)
20080815__bug12921.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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(closes issue #13255)
Reported by: falves11
Patches:
13255-socketport.diff uploaded by qwell (license 4)
Tested by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12170)
Reported by: pj
Patches:
20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36)
Tested by: bbryant, pabelanger
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invite.
(closes issue #11843)
Reported by: pestermann
Patches:
20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36)
20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36)
Tested by: pabelanger
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Diversion: header.
(closes issue #13195)
Reported by: woodsfsg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines
We need to make sure to null-terminate the "name"
portion of SIP URI parameters so that there are no
bogus comparisons.
Thanks to bbryant for pointing this out.
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- Use ast_strlen_zero in one place
- check for successful string comparison the way most of Asterisk code does it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines
Fix rtautoclear and rtcachefriends
(Closes issue #12707)
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the config sample file. Also fix "sip show settings" for tcp connections.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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pointer...
Thanks, eliel!
Reported by: eliel
Patches:
chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64)
(closes issue #13135)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/trunk
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Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
Modified:
branches/1.4/channels/chan_sip.c
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
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a tcpbindaddr if it was disabled.
(closes issue #13117)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
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(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
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(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines
Don't hangup the call if we can't resolve the Contact if there's a proxy
route set for the call.
----
This comment was added a while ago and today it hit me badly.
/* OEJ: Possible issue that may need a check:
If we have a proxy route between us and the device,
should we care about resolving the contact
or should we just send it?
*/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines
Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably.
Reported by: johan
Patches:
12746.txt uploaded by oej (license 306)
Tested by: johan
(issue #12746)
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a peer reference is not leaked.
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and "tlsbindaddr".
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
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- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
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- Set global_outboundproxy.force at reload.
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- Doxygen changes
- Replacing a doxygen description that was copied from another function
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come at a later stage,
as indicated by previous commit message
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problem with this
code, but not the correct problem. Thanks, anyway! ;-)
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sometime in the
future, but also want to avoid compiler warnings now.
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...trying to get my head around the thoughts behind the TCP/TLS stuff
and figure out what needs to be done to make it useful...
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TLS needs to have one configuration per configured domain at some point.
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