Reported by: kenw
Patches:
9724.txt uploaded by russell (license 2)
Tested by: kenw, russell
Resolve a deadlock that occurs when doing a SIP transfer to parking.
I come across this type of deadlock fairly often it seems. It is very important
to mind the boundary between the channel driver and the core in respect to the
channel lock and the channel-pvt lock. Channel drivers lock to lock the
pvt and then the channel once it calls into the core, while the core will do
it in the opposite order. The way this is avoided is by having channel drivers
either release their pvt lock while calling into the core, or such as in this
case, unlocking the pvt just long enough to acquire the channel lock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: casper
Patches:
chan_sip.c.80129.diff uploaded by casper (license 55)
Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it to ever be that value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: jesselang
Patches:
chan_sip-ChannelReload-20080825.patch uploaded by jesselang (license 202)
Remove an extra \r\n to make the ChannelReload event conform with every other event.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
limit on how many history entires will be stored for each SIP dialog. It is
currently set to 50, but can be increased if deemed necessary.
(closes issue #10421, closes issue #10418, patches suggested by jmoldenhauer,
patches updated by me)
(Security implications documented in AST-2007-020)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to the scheduler to ensure that they don't overwrite the ID of a previously
scheduled item. If there is one, it should be removed.
(closes issue #10391, closes issue #10256, probably others, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: irroot
Patches:
sip_timeout.patch uploaded by irroot (license 52)
Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the
IP address.
In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites.
(closes issue #10274, reported by cstadlmann, patched by me with approval from file)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: julianjm
Patches:
chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99)
Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: elandivar
It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@75621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines
(closes issue #10125)
Reported by: makoto
Patches submitted by: makoto
This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the state of a device changes, the device state thread tells the extension
state handling code that it changed. Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent. However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use
the sip_pvt lock wrappers by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
outside of the block that checks for FLAG_FATAL to be set. This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted. A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered. This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235, patch from jcmoore)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
