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${ noResults }
4904 Commits (1c72a8ab495aa9eb658b0afc0041dcb9babd56e7)
Author | SHA1 | Message | Date |
---|---|---|---|
|
5be00864ee |
Prevent loss of Caller ID information set on local channel after masquerade.
Caller ID set on the channel before a masquerade occurs when using a local channel would cause the information to be lost. The problem was that the information was set on a channel destined to be hung up. The somewhat confusing fix is to detect if any Caller ID has been set on the channel and if so preswap the Caller ID data so that basically the masquerade puts the data back. (closes issue #17138) Reported by: kobaz Review: https://reviewboard.asterisk.org/r/847/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@281390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
4f90553a45 |
chan_sip: fixes provisional keepalive scheduled item crash
There is a scheduler item in chan_sip that keeps sending the last provisional message in response to an INVITE Request for a period of time until a final response to that INVITE is sent. Because of the way this scheduler item works, it requires a reference to a sip_pvt pointer to work properly. The problem with this is that it is currently possible (but rare) for the sip_pvt to get destroyed and that scheduler item to still exist. When this occurs, the scheduler event fires and attempts to access a freed sip_pvt which causes a crash. (closes issue #17497) Reported by: anonymouz666 Patches: keepalive_diff_1.4_v2.diff uploaded by dvossel (license 671) Review: https://reviewboard.asterisk.org/r/849/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@281185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
996a394a7d |
Prevent DAHDI channels from overriding the callerid, once it's been set by the user.
(closes issue #16661) Reported by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jstapleton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@280811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7bc66e1d77 |
Backport fixes for sip_uri_params_cmp() from trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8f60bcf42f |
DNID does not get cleard on a new call when using immediate=yes with ISDN signaling.
When you are using chan_dahdi ISDN signaling with immediate=yes and a call comes in without a DNID then you get the DNID of a previous call. Chan_dahdi does not touch the DNID field on a new call if it does not have a DNID. Made always copy the DNID from the new call. The patches backport the relevant changes from trunk -r210387. (closes issue #17568) Reported by: wuwu Patches: issue17568_v1.4.patch uploaded by rmudgett (license 664) issue17568_v1.6.2.patch uploaded by rmudgett (license 664) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b48d6c3dbe |
Regression with T.38 negotiation
Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
1c8d76145b |
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
308d812403 |
priexclusive in chan_dahdi.conf ignored when reloading dahdi module
During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
dca914c43b |
Fix logging message for stale nonce.
(closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8c1f4940eb |
Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
79d795c383 |
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b0202ce6d1 |
Command 'stop gracefully' doesn't.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
14550f93d0 |
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
97f61244f9 |
Fix various typos, reported by Lintian
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
65458abbe8 |
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2cc343faa2 |
Decode URI in contact header of 302 response.
ABE-2352 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
fac72f08b8 |
ss_thread calls pri_grab without lock during overlap dial
Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7eabbade0a |
Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct.
(closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2d31f18ae9 |
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
(closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
5cf7c69afb |
Eliminate deadlock potential in dahdi_fixup().
(This is a backport of 269307, committed to trunk by rmudgett.) Calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. (closes issue #17261) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
25a613a993 |
Need to lock the agent chan before access its internal bits.
Pointed out by russellb on asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
3245bb4b86 |
fixes chan_iax2 race condition
There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
21bdb89a2b |
Merged revisions 270658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r270658 | twilson | 2010-06-15 15:18:04 -0500 (Tue, 15 Jun 2010) | 20 lines Make contactdeny apply to src ip when nat=yes chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270724 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7d953183e6 |
fixes FXS port still ringing when answered, as reported by Tzafrir on dev-list.
(issue #17067) Reported by: tzafrir Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a8f624dc93 |
Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@269495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8427de3128 |
Comment out a rule that likes to break the chan_h323 build randomly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@268053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b3076dd0f5 |
Fix formatting issue with previous patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
abc4bceec2 |
Missing fallback to audio fax feature when T.38 re-INVITE failed
When a T.38 re-INVITE failed with an 488 or 606 answer, we should fallback to audio fax by send a re-re-INVITE without T.38. The function is backported from 1.6 asterisk. (closes issue #16795) Reported by: vrban (closes issue #16692) Reported by: vrban Patches: t38_fallback_to_audio_v3.patch uploaded by vrban (license 756) Tested by: lmadsen, vrban, haggard https://reviewboard.asterisk.org/r/514/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
9d466eed00 |
add dahdi_func_write to zap_tech structure
This was supposed to be committed with r263292, the back-port of teh DAHDI buffer policy dial string option git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ff8057a92a |
fixes build issue with zaptel
(closes issue #17394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@265613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0e37a19b92 |
backport of DAHDI buffer policy dial string option
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
07dbb0f074 |
Registration fix for SIP realtime.
Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@261274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
87e8d0ec46 |
Ensure channel state is not incorrectly set in the case of a very early answer.
The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260434 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
e0d8abbfe2 |
DTMF CallerID detection problems.
The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260195 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2077fcfc6d |
resolves deadlocks in chan_local
Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0ee6131402 |
DAHDI "WARNING" message is confusing and vague
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
e0623c7080 |
hidecalleridname parameter in chan_dahdi.conf
Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@259270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
244f519a25 |
Don't recreate peer, when responding to a repeated deregistration attempt.
When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@257467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
f4fc6dfb4a |
DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock.
SWP-1231 ABE-2163 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@256225 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
e8f9c77145 |
Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel()
(closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@256014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
c689d464dd |
Don't kill Asterisk if the H323 listener does not start.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@255409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
43393bd1a9 |
Resolve a number of FreeBSD build issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@253631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
1b2dbf6c1a |
Revert API change in release branches
This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
529e8af144 |
Merged revisions 252089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a247e69d65 |
Forward declaring dahdi_pri was already done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@251997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0bc0edcda9 |
Make chan_dahdi wakeup_sub() prototype not conditional.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@251986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
3896fecd58 |
Make sure to clear red alarm after polarity reversal.
From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@250480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8dbeb5925d |
fixes problem with duplicate TXREQ packets
When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@250394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
9a4e4eb749 |
Modify queued frames from local channels to not set the other side to up
In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@249536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
271908886f |
overlap receiving: automatically send CALL PROCEEDING when dialplan starts
Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis (closes issue #16789) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@249365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |