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${ noResults }
5767 Commits (4cc1606d2708c9435e3e5726891b7882e5d62660)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
b605bdff0c |
Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
e70fc3d136 |
Broke up the large conditional blocks so it is easy to see if a function is compiled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
326587ebe3 |
Fix a spelling mistake.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a9ee30da54 |
Add a todo. I do need to really check what's going on with this kill-the-user business ;-)
Why do we suddenly have two flags to set peer type? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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81a3d40c08 |
Small formatting change
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9ea148b260 |
Add some well-needed improvements to the wishlist in the code, so that we can close
some bug reports. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2af8f59958 |
The CID lookup feature wasn't actually working properly with dialog-info+xml
supporting devices. The devices (snoms, specifically) need to receive a SIP URI instead of just an extension. This adds that functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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97b4e9cf2a |
channels/chan_dahdi.c
* Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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efd6586546 |
Revert two lines that was extra, but only on fridays.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172319 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7ecda45482 |
Fix "cancel answered elsewhere" through app_queue with members in chan_local.
Also, implement a private cause code (as suggested by Tilghman). This works with chan_sip, but doesn't propagate through chan_local. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b79a12e929 |
- Make sure we set setvar= variables on outbound calls too, not only inbound calls.
- Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3e144e2a71 |
Make sure register= line supports both port and expiry at the same time.
(closes issue #14185) Reported by: Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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55782a8dfa |
Merged revisions 172169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f029e410ec |
A further correction: cast the sizeof to an int.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5918a38b98 |
my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 didn't like the \%ld and issued a warning, breaking my dev-mode build. This fixes it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ca052f64d0 |
Merged revisions 171963 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7a69506727 |
Don't complain about lack of D-channels on PTMP connections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5f1a4ebe6d |
Merged revisions 171689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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906d665477 |
Handle new VMWI ioctl structure (Now there are two VMWI ioctl calls.)
(issue #14104) Reported by: alecdavis Tested by: dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a0a8a4d68e |
Solving the same issue, but a bit different in trunk...
Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e9beff5969 |
Moving generic setting to friends
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171401 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f35874c3f5 |
Continue to move variables into the sip_cfg structure to make them easier to handle in the future as a group of settings for a group of devices.
At some point, I want one sip_cfg per domain handled, so we can have "group" settings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a6228ccaf3 |
Just moving around variable declarations so that we have all globals in the same place.
Default setting is set before we activate the channel or at reloads, not where we declare the variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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08640496d1 |
Merged revisions 171264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171326 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b0a29390ec |
Merged revisions 171187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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630fe3ccb8 |
dont segfault when a MWI event occurs on a line without a registered device
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0308c5b943 |
Merged revisions 170588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 Jan 2009) | 2 lines Additions to AST-2009-001 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a394fe3a7a |
Merged revisions 170504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold. (closes issue #14295) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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40bfad1212 |
let's use SENTINEL where needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f5ee4e99fa |
We should not do restart messages if we're in PTMP mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a992164529 |
Dont clear the display of skinny phones when not needed.
(closes issue #13182) Reported by: pj Patches: 2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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de63e93a39 |
MWI messages included in CID spill was not being properly handled and prevented the call from being processed
(issue #14313) Reported by: seandarcy Tested by: dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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55fd5f2d2d |
Make sure we don't set the channel to be inalarm for a D-channel drop on PTMP connections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9a28a07739 |
change VMWI to use new DAHDI_VMWI ioctl call.
Change configure script to detect the new ioctl call data structure. (issue #14104) Reported by: alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded by dbailey (license ) Tested by: alecdavis, dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2da6670f10 |
* Adjust some conditionals to balance curly braces.
* Other minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e13016057e |
Whitespace changes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f6bb99a5c5 |
Test commit for test issue #14303
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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28dc82ea4e |
Further fix some oddities in sip show users and sip show peers logic
ccesario on IRC pointed out that his sip peers were not displayed properly when he would issue the command "sip show peers." The problem was that the onlymatchonip field was used to determine if the endpoint was a "peer" or "user." The tricky part is that a "friend" is supposed to be treated as both a "user" and a "peer" but the logic would not allow "friends" to show up as "peers" since onlymatchonip was set to FALSE for friends. I have modified the sip_peer structure to more explicitly keep track of what type endpoint it is so that the various manager and CLI commands will display the expected information Reported by ccesario via IRC Tested by ccesario git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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419bb42e04 |
Convert the character pointers in a sip_request to be pointer offsets
When an ast_str expands to hold more data, any pointers that were pointing to the data prior to the expansion will be pointing at invalid memory. This change makes such pointers used in chan_sip.c instead be offsets from the beginning of the string so that the same math may be applied no matter where in memory the string resides. To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so that given a sip_request and an offset, the string at that offset is returned. (closes issue #14220) Reported by: riksta Tested by: putnopvut Review http://reviewboard.digium.com/r/126/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1d781feca6 |
Redo the event-based MWI in chan_skinny.
Dan saw regular segfaults with the old implementation and rewrote it to make it really eventbased. I altered it to be trunk compatible and wedhorn gave some feedback and ideas how to make it even better. (closes issue #13821) Reported by: DEA Patches: chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by: mvanbaak, DEA "no probs by me" from wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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42d4853f61 |
Get rid of magic number and replace with DAHDI_VMWI_NUMBER_MASK when
determining the number of messages pending for MWI call git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169325 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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82d76adeb8 |
Add enhanced MWI generation to take advantage of new dahdi line reversal MWI ability.
(closes issue #14104) Reported by: alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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231e7efbe5 |
Merged revisions 169210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a potential NULL pointer dereference Move the check for if both channels on a local_pvt have generators to below where p->chan is checked for NULLity (NULLness?). This prevents a crash from occurring if p->chan is NULL. (closes issue #14189) Reported by: sascha Patches: 14189.patch uploaded by putnopvut (license 60) Tested by: sascha ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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65120a3b33 |
Add discriminator for when ring pulse alert signal is used to preface MWI spills
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169153 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6b32f4600d |
Fix port :0 added to SIP INVITE URI when outboundproxy used
(closes issue #14233) Reported by: chris-mac Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a2ac43cb8a |
Merged revisions 168975 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines Account for possible NULL pointer when we receive a 408 in response to a REGISTER It may be that by the time we receive a reply to a REGISTER request, the attempt has timed out and thus the registry structure pointed to by the corresponding sip_pvt has gone away. This situation was handled properly for a 200 OK response, but the 408 case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash This commit fixes this assumption and prints out a message to the console if we should receive a late 408 response to a REGISTER (closes issue #14211) Reported by: aborghi Patches: 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168976 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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de535452cd |
build in dev mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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73694e15f1 |
Add missing brace
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168732 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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78c440b7ad |
Fix the compactheaders option in sip.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5c9daf4d3d |
Remove an unneeded condition for line addition to a SIP request/response
In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically allocated buffer to hold the text of the request. There was a check in the add_line function to not attempt to write the line into the buffer if we did not have room for it. In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str structure is used to hold the text. Since it may grow to fit an arbitrarily sized string, this check in add_line is no longer valid. I found this oddity while attempting to fix issue #14220; however, I do not believe that this is the fix for that issue since the output supplied by the reporter did not contain the warning message that would be printed had this condition been satisfied. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168725 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |