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5788 Commits (4ab40e9a10ab8cb1409e28703f50b89868d07afc)
Author | SHA1 | Message | Date |
---|---|---|---|
|
35ac1d7e1c |
Fixed iax2 key rotation backwards compatibility
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. Review: http://reviewboard.digium.com/r/159/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
2a53f2ec98 |
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d11e6f0591 |
Add dynamic fax buffer configuration option to chan_dahdi.conf
When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a741658f58 |
Remove useless string copy, and make sscanf safe again
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
178e6f06df |
Adds force encryption option to iax.conf
This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well. (closes issue #13285) Reported by: sgofferj Tested by: russell Review: http://reviewboard.digium.com/r/150/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
768c73160e |
Avoid using ast_strdupa() in a loop.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1448b5db6a |
correct warning message to not refer specifically to DAHDI
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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53f3ab973e |
Setting key rotation to be off by default
Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c873d72ca2 |
Issue a warning message if our candidate's IP is the loopback address.
(closes issue #13985) Reported by: jcovert Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7bb5ef8399 |
Merged revisions 175029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6304c09149 |
Only decrease inringing count if above zero.
(issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
8e6780a5b1 |
Set the type for the peer structure to be a peer as the default.
(closes issue #14447) Reported by: triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bb327036f1 |
Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
(closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e0b0ae07a3 |
Fix something I messed up in the merge I just did
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a02ef73b25 |
Merged revisions 174282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0024ad62ab |
Merged revisions 174082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
c15b83e7e5 |
Adds immediate yes/no option to iax.conf
This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well. (closes issue #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk uploaded by jcovert (license 551) iax.conf.sample.patch uploaded by jcovert (license 551) Tested by: jcovert, dvossel Review: http://reviewboard.digium.com/r/143/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fa389ab17a |
Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322) Reported by: amessina git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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550f7f1e65 |
Merged revisions 173967-173968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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647b68ec23 |
Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0cbada4f96 |
Add a common implementation of a scheduler context with a dedicated thread.
This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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 |
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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 |
|
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 |
|
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 |