mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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10.7.0
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10.7.0-digiumphones-rc1
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10.8.0
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10.9.0
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13.0.0
13.0.0-beta1
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13.0.1
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13.1-cert2
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13.1.1
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13.3.1
13.3.2
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13.7.1
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certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
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certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
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certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
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certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.2-cert2
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certified/11.6-cert1
certified/11.6-cert1-rc1
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certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
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certified/11.6-cert6
certified/11.6-cert7
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certified/11.6-cert9
certified/13.1-cert1
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certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
1272 Commits (6aaa119e84befc2b05bfeff5437f033dbca8d783)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Matthew Nicholson
|
7ed425ec80 |
This patch adds a sequence field to CDRs that can be combined with the linkedid or uniqueid field to uniquely identify a CDR.
(closes issue #15180) Reported by: Nick_Lewis Patches: cdr-sequence10.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
2263ced9dd |
Add support for using a hint when configuring a state interface using the format hint:<extension>@<context>.
(closes issue #15168) Reported by: p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Leif Madsen
|
db78eeb7c7 |
Additional fixes to the extensions.conf.sample file.
Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
f6827928b0 |
Update extensions.conf.sample file to fix incorrect extensions.
(closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
66579d9d49 |
Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks
(closes issue #12950) Reported by: alea-soluciones Patches: ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514) Tested by: alea-soluciones, adomjan, urtho, nahuelgreco git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Matthew Nicholson
|
93e43578ec |
This patch adds support for a draft proposal for adding Q.850 reason headers to sip messages.
(closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
5524f0ab11 |
Merged revisions 226382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
5825f68e8b |
Add support for receiving unsolicited MWI NOTIFY messages.
This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
|
cff6d02b53 |
Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
17 years ago |
|
Tilghman Lesher
|
d9f72c1893 |
Permit storage of voicemail secrets in a separate file, located within the spool directory.
(closes issue #14276) Reported by: klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65) Tested by: jamesgolovich git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
01ab66275a |
Add support for specifying the IP address to use for media streams in sip.conf
This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
a31eb5bb35 |
Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
984d6500ce |
Merged revisions 225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
28d0ec5421 |
Add support for specifying the IP address to use for media streams in sip.conf
(closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Matthew Nicholson
|
26638d3a55 |
Add dynamic range compression support for analog channels.
(closes issue AST-29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
97bf6e881a |
Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Doug Bailey
|
fb1433f43f |
chan_dahdi.conf.sample changes for DTMF CID detect
Explains new options for detecting DTMF CID on fxo lines (issue #9096) Reported by: fleed Patches: chan_dahid_sample_config.patch uploaded by sum (license 766) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
e3f473f4f3 |
Allow for adding message body to the SIP NOTIFY message
Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
b14857f49d |
Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options
SWP-151 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
|
fb41713f99 |
Adding note about TLS usage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
5c1f05576c |
Add an additional note on TLS support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
0224b47994 |
Adding some information on TLS support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
|
d4bd570985 |
Remove 'keepstats' queue option from sample config, as it's no longer used.
https://reviewboard.asterisk.org/r/115/ (closes issue #15820) Reported by: kshumard git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
1d40faebac |
contact header port ignored transport when using externip
This patch adds support for TCP/TLS in the Contact header when using NAT, specifically externip or externhost. The original issue was that Asterisk sent 5060 as the port in the contact header whether TLS was used or not. Additionally, this patch adds 2 config options to sip.conf, specifically externtcpport and externtlsport. This allows a user to specify different external ports for TCP and TLS other than those used internally, this is especially useful in in a PAT/port redirection setup. Thanks to ebroad for reporting the issue and providing the patch! (closes issue #15880) Reported by: ebroad Patches: portmap.patch uploaded by ebroad (license 878) externtXXport_v2.patch uploaded by ebroad (license 878) Tested by: ebroad Review: https://reviewboard.asterisk.org/r/392/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222398 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
|
20743ec07d |
Allow non-compliant T.38 endpoints to be supportable via configuration option.
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Kevin P. Fleming
|
19ba91cd22 |
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Matthew Nicholson
|
a5eee590f4 |
Merged revisions 221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221432 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthias Nick
|
63984d5c21 |
Merged revisions 221153,221157,221303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
|
865daf4858 |
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Philippe Sultan
|
b11b94a083 |
Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
79b9b75eab |
Documentation in the commit messages is soon forgotten, please add it to the docs in the product.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
c68a2d9d30 |
Add support for 'setvar=' for MGCP device lines, like other channel drivers provide.
(closes issue #14818) Reported by: alea-soluciones Patches: chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
3093ccb619 |
Merged revisions 219023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
a873ad7a9b |
Recorded merge of revisions 218331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
75d8960740 |
Allow multiple rows to be fetched within the normal mode of operation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
8af3a908a9 |
Update sip.conf.sample documentation, reorganize a bit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
98f18d56b8 |
Merged revisions 216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
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d09f9fd00a |
Merge code associated with AST-2009-006
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Richard Mudgett
|
595ab444af |
Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ISDN PTMP CPE spans.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Richard Mudgett
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e6d5478a50 |
Minor punctuation change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jason Parker
|
8942f4e4a1 |
Merged revisions 213493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | 5 lines Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213494 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
3028e257bb |
Better parsing for the "register" line
Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
97d93fbfca |
Make the default extconfig.conf match entries with the sample res_mysql.conf.
This eliminates a future source of possible confusion with the configuration of 1.6.1 and higher. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Nicholson
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5583a4e955 |
This patch adds support for choosing a realm based on the domain in the From or To header in the incoming request. Eligible domains are taken from the domains list in the config file. This functionality is enabled when domainsasrealm is enabled in the config file.
(closes issue #11361) Reported by: arkadia Patches: sip_realm_mnich_to_added_2.patch uploaded by arkadia (license 233) Tested by: arkadia git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
|
e9d22f802e |
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
17 years ago |
|
Matthew Nicholson
|
d0664ba6af |
Add an 'sms' option to mobile.conf to manually enable or disable SMS support.
(closes issue #15071) Reported by: ughnz Patches: optional-sms1.diff uploaded by mnicholson (license 96) Tested by: ughnz, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
c058252718 |
Add configuration sample code for previous commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
ba8dcde549 |
Merged revisions 209131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Michiel van Baak
|
126bf8eeb5 |
add default alias reload to run module reload.
Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jeff Peeler
|
496b509c42 |
Update some missing allowed options for overlapdial
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |