mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
1598 Commits (dec9ee2f042bcddb38a9393c95a7a79e4cb2e2c2)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Brent Eagles
|
ab894d5af9 |
This change adds a SIP peer configuration feature to allow the peer's
configured codecs to take precedence on an outgoing call. This change introduces a new peer configuration property named 'ignore_requested_pref' that causes the requested codec to be ignored when determining the preferred codec for an outgoing call leg. The consequence is that Asterisk's usual efforts to prefer avoiding transcoding can be overridden on a peer-by-peer basis where appropriate. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Joshua Colp
|
898ca023d5 |
Tweak extension used for incoming calls received on Motif.
Based on feedback from numerous individuals this patch tweaks incoming calls to first look for an extension with the name of the endpoint. If no such extension exists the call will silently fall back to the "s" extension as it previously did. ........ Merged revisions 376983 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
866d968149 |
Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
With ICE support enabled in chan_sip and a large number of interfaces on the system it was possible for the produced SDP to be truncated due to some fixed size buffers. These buffers have now been changed so they will dynamically grow as needed. ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is no longer enabled by default there. (closes issue ASTERISK-20643) Reported by: coopvr ........ Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jonathan Rose
|
d4a357b82f |
chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry
A regression was introduced in chan_sip by changes to sip reload introduced by r349097. That patch moved peer purging from the beginning of the reload to after the general configuration was finished. This patch fixes that by undoing the repositioning of the original peer purging code and using a similar function after performing general configuration that purges only autocreated peers that were created when persist mode isn't enabled. (closes issue ASTERISK-20611) Reported by: Alisher Review: https://reviewboard.asterisk.org/r/2171/ ........ Merged revisions 375575 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Pedro Kiefer
|
8b34dc8192 |
Adds new formats to app_alarmreceiver, ALAW calls support and enhanced protection.
Commiting this on behalf of Kaloyan Kovachev (license 5506). AlarmReceiver now supports the following DTMF signaling types: - ContactId - 4x1 - 4x2 - High Speed - Super Fast We are also auto-detecting which signaling is being received. So support for those protocols should work out-the-box. Correctly identify ALAW / ULAW calls. Some enhanced protection for broken panels and malicious callers where added. (closes issue ASTERISK-20289) Reported by: Kaloyan Kovachev Review: https://reviewboard.asterisk.org/r/2088/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375150 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Richard Mudgett
|
3e1d2917bb |
dahdi.conf.sample: Add description for "buffers" setting.
This contains an edited version of the patch originally created by John
Bigelow.
(closes issue ASTERISK-14435)
Reported by: John Bigelow
Patches:
buffers.patch (license #5091) patch uploaded by John Bigelow
0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell
Modified
........
Merged revisions 374727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 374728 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 374729 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Matthew Jordan
|
3620fcff36 |
Disable ICE support by default
Since there are a number of legacy devices out there that fail to handle ICE candidates properly (which is a nice way of saying something much uglier), disable it by default. Support for ICE candidates can be enabled in rtp.conf using the icesupport setting. ........ Merged revisions 374676 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Alec L Davis
|
90f8c90b10 |
dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END
Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ ........ Merged revisions 374479 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374481 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374485 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Alec L Davis
|
4af961a03a |
dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ ........ Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374385 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374386 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
2b56626b43 |
Remove dead code and documentation for nonexistent feature.
multiplelogin was removed from chan_agent back in 1.6.0 when AgentCallbackLogin() was removed. (closes issue AST-948) reported by Steve Pitts ........ Merged revisions 373768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373769 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373770 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Terry Wilson
|
b7233b18eb |
Properly handle UAC/UAS roles for SIP session timers
The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373665 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kinsey Moore
|
d7085e431f |
Fix documentation for default username in res_odbc
This was previously stated to be "root", but is actually the name of the context if unspecified. (closes issue ASTERISK-20258) Reported by: Stefan x ........ Merged revisions 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373579 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373580 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Brent Eagles
|
f787f4219a |
res_rtp_asterisk: Make TURN and STUN server configurations consistent.
This patch removes the turnport configuration property and changes the turnaddr property to be a combined host[:port] configuration string. The patch also modifies the documentation in the example configuration to reflect the property changes and adds some additional text indicating how the STUN port is configured. (closes issue ASTERISK-20344) Reported by: beagles Tested by: beagles Review: https://reviewboard.asterisk.org/r/2111/ ........ Merged revisions 373403 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
ca0e96ae19 |
Add queue monitoring hints
This patch adds support for hints on a queue. Hints can be added using the nomenclature 'Queue:name', where name is the name of the queue being monitored. This nifty feature was done by Alec Davis. Review: https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis Tested by: alecdavis patches: review1619.diff2 by alecdavis (license 585) ........ Merged revisions 373235 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
e8380afc8a |
Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
As mentioned on the review for this, WebRTC has moved towards choosing DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds support for this but makes it available for normal SIP clients as well. Testing has been done to ensure that this introduces no regressions with existing behavior and also that it functions as expected. Review: https://reviewboard.asterisk.org/r/2113/ ........ Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Alec L Davis
|
67ca3b9126 |
app_queue: Support an 'agent available' hint
Sets INUSE when no free agents, NOT_INUSE when an agent is free. modifes handle_statechange() scan members loop to scan for a free agent and updates the Queue:queuename_avial devstate. Previously exited early if the member was found in the queue. Now Exits later when both a member was found, and a free agent was found. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2121/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
1a79c9c182 |
logger: Add rotatestrategy option of 'none' which does not perform rotations
With this option in use, it may be necessary to regulate your log files
externally.
(closes issue ASTERISK-20189)
Reported by: Jaco Kroon
Patches:
asterisk-logger-norotate-trunk.patch uploaded by Jaco Kroon (license 5671)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Darren Sessions
|
7e46e4d17b |
LDAP Realtime Peers Cannot Register
Prior to 1.8, it was not necessary for an explicit "type" to be set for an asterisk LDAP realtime peer. Now the routine find_peer actually checks the type field during registration and fails to find the peer if it is not set. The attached patch makes the realtime type equal whatever type is being searched for if the type is 0 upon return from routine build_peer. (closes issue ASTERISK-17222) Reported by: John Covert Patch by: David Vossel Tested by: Darren Sessions Review: https://reviewboard.asterisk.org/r/2095/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
44d854938b |
Fix misleading documentation in agents.conf.sample regarding ackcall usage.
The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts ........ Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371789 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371790 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
4b50e3f1ee |
Fix incorrectly documented option in queues.conf
sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts ........ Merged revisions 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371748 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371750 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
5c4578f4ad |
Add named callgroups/pickupgroups
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
4377d511ae |
Add headers from SIPAddHeader to outbound REFER requests.
This is a patch from kkm from review board. This is useful for adding headers to REFER requests that emanate from a Transfer() dialplan application call. This also fixes some uses of the Referred-by header, removing an extra set of angle brackets. I've modified the reporter's original patch to not require any additions to the sip_refer header and to just remove the referred_by_name from sip_refer since it is no longer needed or used. (closes Issue ASTERISK-17639) reported by Kirill Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff uploaded by Kirill Katsnelson (license #5845) Review: https://reviewboard.asterisk.org/r/1159 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
58f281a670 |
Add "setvar" option to manager.conf.
With this option set, channel variables can be set on every manager originate. The Variable header can still be used to set additional channel variables for individual calls if desired. This work was completed by Olle Johansson on review board. I have applied the review feedback and am committing it in order to get this into trunk before Asterisk 11 is branched. Review: https://reviewboard.asterisk.org/r/1412 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
a28e6fc7bd |
Add separate configuration options for subscription and registration minexpiry and maxexpiry.
This offers more fine-grained control over how long subscriptions last without negatively affecting the expiration range for registrations. Uploaded by: Guenther Kelleter(license #6372) Review: https://reviewboard.asterisk.org/r/2051 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370386 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
4d6b524b61 |
Prevent multiple local candidates from being added with the same information and add support for disabling ICE on a per-peer basis.
(closes issue ASTERISK-20088) Reported by: wimpy Review: https://reviewboard.asterisk.org/r/2044/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
ded09e3682 |
named_acl: Remove systemname option from acl.conf, use asterisk.conf value
Review: https://reviewboard.asterisk.org/r/2057/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Igor Goncharovskiy
|
9278b5e51e |
Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant.
|
13 years ago |
|
Joshua Colp
|
e938737570 |
Add support for SIP over WebSocket.
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
a7dfafdc56 |
Handle deprecated (aliased) option names with the config options api
Add a simple way to register "deprecated" option names that alias to a different "current" name. Review: https://reviewboard.asterisk.org/r/2026/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370043 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
10afdf3a2a |
Named ACLs: Introduces a system for creating and sharing ACLs
This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
04504e80a3 |
Document that multiple endpoints using the same connection is not supported.
(closes issue ASTERISK-20104) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jason Parker
|
b1cd995273 |
Add Digium phones context to sip_notify sample config.
This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369819 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
a3fa37b8cf |
Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif.
This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either. These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold, unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications. The original Google Talk protocol is also supported for Google Voice interoperability. You may ask yourself though where the name motif comes from... and I would say to you... music! motif: a perceivable or salient recurring fragment or succession of notes Sorta like a jingle! Review: https://reviewboard.asterisk.org/r/1917/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Alexandr Anikin
|
fa10f3f8a8 |
Added direct media support to ooh323 channel driver
options are documented in config sample sample config rename to proper name - ooh323.conf To change media address ooh323 send empty TCS if there was completed TCS exchange or send facility forwardedelements with new fast start proposal if not. Then close transmit logical channels and renew TCS exchange. If new fast start proposal is received then ooh323 stack call back channel driver routine to change rtp address in the rtp instance. If empty TCS is received then close transmit logical channels and renew TCS exchange Review: https://reviewboard.asterisk.org/r/1607/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
213bbc169a |
Add a cleaned up drop-in replacement for res_jabber called res_xmpp. This provides the same externally facing functionality but is implemented differently internally.
This is currently not built by default but this will be changed once chan_jingle2 (insert actual name in your head when reading this after it has been merged) is in the tree. Review: https://reviewboard.asterisk.org/r/1983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
37256ea45d |
Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
Review: https://reviewboard.asterisk.org/r/1891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
ac35b92b62 |
Hangup handlers - Dialplan subroutines that run when the channel hangs up.
Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
1609fca6bb |
Add the ability to set flags via the config options api
Allows the setting of flags via the config options api.
For example, code like this:
#define OPT1 1 << 0
#define OPT2 1 << 1
#define OPT3 1 << 2
struct thing {
unsigned int flags;
};
and a config like this:
[blah]
opt1=yes
opt2=no
opt3=yes
Review: https://reviewboard.asterisk.org/r/2004/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Mark Michelson
|
14a985560e |
Merge changes dealing with support for Digium phones.
Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
6016094db7 |
Add missing config for config API test
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368221 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
d54717c39e |
Add new config-parsing framework
This framework adds a way to register the various options in a config file with Asterisk and to handle loading and reloading of that config in a consistent and atomic manner. Review: https://reviewboard.asterisk.org/r/1873/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368181 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
463f9d729a |
Help mitigate potential reinvite glare scenarios.
When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. Review: https://reviewboard.asterisk.org/r/1954 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
ec3b8a1f27 |
app_queue: Per Member ringinuse option and deprecation of ignorebusy
Adds a number of methods for controlling the setting of 'ringinuse' which is basically the same concept as the old ignorebusy setting, only now the per member setting always controls whether or not the member is actually ringed while in use. A CLI command and a manager action have been added to change a given queue member's ringinuse option while Asterisk is running and the an argument has been added for adding members with deliberately set ringinuse in queues.conf Some effort has been made to ensure compatability with dialplans and databases still referring to 'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1919/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
b5a6de76fc |
Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
108f5fafd7 |
Improve FollowMe accept/decline DTMF string matching.
If you hit the wrong DTMF digit trying to accept/decline a FollowMe call, you had to wait for the prompt to repeat to try again. * Make FollowMe compare the last DTMF digits received to the accept/decline matching strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
d71d8ed995 |
Keep answered FollowMe calls until call accepted or last step times out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
|
ae1502be33 |
Add support for lightweight NAT keepalive.
If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Olle Johansson
|
7aa0c3c64b |
Make it possible to change the minimum DTMF duration in asterisk.conf
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Richard Mudgett
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47ccc7f5d6 |
Update membermacro and membergosub documentation in queues.conf.sample.
........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362678 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Richard Mudgett
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c7cb03a975 |
Add ability to ignore layer 1 alarms for BRI PTMP lines.
Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |