mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
7298 Commits (11bbc097b7d6e115ea413dd80b8deee95ee3dc63)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
2736fe9917 |
Defer sending the connected line reinvite if a reinvite is already in progress.
(issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) ........ Merged revisions 358162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358163 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358164 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
dec0d4f9e3 |
Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Reported by: Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358116 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
b71deb0518 |
Fix unused-but-set-variable warnings
All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. ........ Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358033 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
e8f8d2c81e |
Make chan_usbradio compile under dev mode
x=++x and x=x=1? Really? ........ Merged revisions 357986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357987 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
9926662aba |
Remove ISDN hold restriction for non-bridged calls.
The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357895 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
ced1211fad |
Fix compile error from latest channel opaquification change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
f6b2f05f8c |
The default value for mohinterpret is the empty string, so when resetting to
default values don't explicitly set the value to "default." ........ Merged revisions 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357812 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
4094a9f57e |
Fix compilation error due to typo during channel opaquification.
s/ast_channel_fd_set/ast_channel_internal_fd_set/g git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
0e5c761c28 |
Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
a9d607a357 |
Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
764d2ccae2 |
Use more reasonable cause code when rejecting incoming call waiting calls.
(closes issue ASTERISK-19397)
Reported by: Birger Harzenetter
Patches:
nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter
........
Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 357408 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
1bef7695ce |
Add a security event for the case where fake authentication challenge is sent.
........ Merged revisions 357318 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357319 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
85ea4277f1 |
Convert struct ast_tcptls_session_instance to finally use the ao2 object lock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
565f411868 |
Changes transport option in sip.conf so that using multiple instances doesn't stack.
Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header ........ Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357271 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c20cfcdcf0 |
Address comments from Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3cf09f40f7 |
Convert netsock.h over to use ast_sockaddrs rather than sockaddr_in and update
chan_iax2 to pass in the correct types. chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk at this point, so this feels like a safe change to make. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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299dd5d4fc |
Adds an option to sip.conf that prevents diversion headers from being added.
send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9ed6de9fd2 |
There isn't much point in saving off and restoring a value that we never use again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356966 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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51c24c88a1 |
Prefer ast_set_qos() over ast_netsock_set_qos()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ebe2c33b72 |
Fix worker thread resource leak in SIP TCP/TLS.
The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356690 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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670797e5da |
Allow SRTP policies to be reloaded
Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ebaf59a656 |
Opaquification for ast_format structs in struct ast_channel
Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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235f88d122 |
Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.
Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356522 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c078a1819c |
Fix ACK routing for non-2xx responses.
When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx final
response to an INVITE, we are supposed to send the ACK to the same place
we initially sent the INVITE.
We had been doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression where we would
use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK based on the
invitestate. If it is INV_COMPLETED then that means that we have received
a non-2xx final response (INV_TERMINATED indicates a 2xx response was received).
If it is INV_CANCELLED, then that means the call is being canceled, which
means that we should be ACKing a 487 response.
The other change introduced here is setting the invitestate to INV_CONFIRMED
when we send an ACK *after* the reqprep instead of before. This way, we can
tell in reqprep more easily what the invitestate is prior to sending the ACK.
(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
(with some slight modifications prior to commit)
........
Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 356476 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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25a9b03cd1 |
Correct some set-but-unused variable warnings in the mISDN library.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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90a6848c67 |
Fix chan_misdn after the lastest opaquification changes
It now compiles, but there are some unrelated warnings for set but unused variables. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a8d9e0bf0b |
Merged revisions 356215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r356215 | mjordan | 2012-02-22 08:53:53 -0600 (Wed, 22 Feb 2012) | 32 lines Merged revisions 356214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is determined, the number of peers in the container can change, as the peers container is not locked. This could cause a buffer overrun when populating the array, if peers were added to the container after the array was created. Additionally, a memory leak of the allocated array would occur if a user caused the _show_peers method to return CLI_SHOWUSAGE. We now create a snapshot of the current peers using an ao2_callback with the OBJ_MULTIPLE flag. This size of the array is set to the number of peers that the iterator will iterate over; hence, if peers are added or removed from the peers container it will not affect the execution of the "sip show peers" command. Review: https://reviewboard.asterisk.org/r/1738/ (closes issue ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1c971ae604 |
Make 'iax2 show callnumber usage' output make sense when an IP is passed in.
........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356108 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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57f42bd74f |
ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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25e5eb3b96 |
Remove spurious warning when 'qualifyfreqnotok' is set successfully.
(closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355998 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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db487bd7f8 |
This was a LOG_NOTICE, so roll it back.
........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355953 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2bd6649a93 |
Change some debug messages from LOG_DEBUG to ast_debug.
........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355950 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bec0ee0851 |
Add some boilerplate documentation for IAXVAR and IAXPEER.
........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355905 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2c1b3144cb |
Set the length of the ast_sockaddr, so that we can set it's port later.
Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355902 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a4f6d96b2e |
push 'outgoing' flag from sig_XXX up to chan_dahdi
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3816fdde94 |
Don't allow trunkfreq to be greater than 1000ms.
........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355794 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7c373d8c13 |
Pass the correct value to ast_timer_set_rate() for IAX2 trunking.
IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355747 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355748 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8a20faa8d7 |
Fix regressions with regards to route-set creation on early dialogs.
This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 ........ Merged revisions 355732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355733 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b69fb773d2 |
When IAX2 debugging is enabled, make sure to log 'apathetic' messages too.
........ Merged revisions 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355530 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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45f361c9bd |
Remove IAX_OLD_FIND from chan_iax2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0d12368261 |
Use TRUNK_CALL_START as originally intended.
Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. ........ Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355449 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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03894236d0 |
Properly invert the return of a strncmp call.
This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) ........ Merged revisions 355268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355271 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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98111f8f1f |
Clear the high order bit from the destination call number before sending.
send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. ........ Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355183 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355184 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d8af1a4882 |
Fix compile error from most recent ast_channel opaquification installment.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355055 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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34c55e8e7c |
Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6225c6cadc |
Fix parsing of SIP headers where compact and non-compact headers are mixed
Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ ........ Merged revisions 354702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354703 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e5c51ee44c |
Add auto_force_rport and auto_comedia NAT options
This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dff9b61f5c |
Clean-up of minor formatting issues in r354542/3/4
rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ ........ Merged revisions 354547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354548 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ba08e9f4d6 |
Fix SIP INFO DTMF handling for non-numeric codes
In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ Merged revisions 354542 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354543 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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16fbc7e902 |
Fix some compile problems from the 'cppcheck' patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |