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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines
Include examples of FILTER usage in extension patterns where a "." may be a risk.
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First off, these options weren't actually doing anything.
By the time the options were parsed, the peer and dialog
containers had already been allocated with their default
values.
Second, hash table size is something that doesn't really
make sense to change in a config file. If a user is that
interested in changing the hashtable size, he can modify
the source itself.
I have removed the parsing of the hash_peer, hash_user,
and hash_dialog options. I have removed the hash_user_size
variable altogether since it is not used at all. I also
changed hash_peer_size and hash_dialog_size to be constant,
and have changed the symbols to be in all caps as constants
typically are. I have also removed the entire section in
sip.conf.sample regarding configurable hashtable sizes.
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functionality and provide commonly useful features.
(closes issue #16090)
Reported by: pprindeville
Patches:
extensions.conf-bugid16090.patch#3 uploaded by pprindeville (license 347)
Tested by: tzafrir, pprindeville, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239834 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16411)
Reported by: stanusr
Patches:
__20091210-sip.conf.sample-documentation.txt uploaded by lmadsen (license 10)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239520 f38db490-d61c-443f-a65b-d21fe96a405b
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(typically, UTC)
(closes issue #16401)
Reported by: lordmortis
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2. Added service type ported number query.
3. Formated code.
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The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235342 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines
Add a line showing that we can use CIDR notation.
patch by jsmith, after discussion with jtodd
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235298 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15132)
Reported by: floletarmo
Patches:
voicemail_changes.patch uploaded by floletarmo (license 784)
(with some additional changes by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234820 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines
clarify requirecalltoken option in iax.sample.conf
(closes issue #16223)
Reported by: bklang
Patches:
clarify-iax-requirecalltoken.patch uploaded by bklang (license 919)
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members when enabled
(closes issue #14559)
Reported by: fiddur
Patches:
trunk-199584-1.diff uploaded by fiddur (license 678)
Tested by: fiddur, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233196 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16263)
Reported by: andrew
Patches:
pagerdate.patch uploaded by andrew (license 240)
(with a slight modification by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232916 f38db490-d61c-443f-a65b-d21fe96a405b
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Previously only possible per context, new option called imapfolder.
(closes issue #14298)
Reported by: jablko
Patches:
patch-200906202 uploaded by jablko (license 675)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232700 f38db490-d61c-443f-a65b-d21fe96a405b
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Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230881 f38db490-d61c-443f-a65b-d21fe96a405b
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(Closes AST-33)
Reviewboard: https://reviewboard.asterisk.org/r/368/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230111 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/426/
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Just updating a spelling error and some capitalization in a
documentation update that Olle added. May the Swenglish be
with you.
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calculating the time of announcments from the end of the previous announcment rather than from the beginning.
(closes issue #15260)
Reported by: tonils
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228947 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@227435 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@227424 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@227361 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15857)
Reported by: pprindeville
Patches:
stdexten.patch#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227162 f38db490-d61c-443f-a65b-d21fe96a405b
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(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: http://svn.digium.com/svn/asterisk/trunk@227049 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@226687 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226384 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@226060 f38db490-d61c-443f-a65b-d21fe96a405b
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* 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: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@225406 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@225089 f38db490-d61c-443f-a65b-d21fe96a405b
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in the next version.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225034 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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(closes issue #14729)
Reported by: _brent_
Patches:
media_address.patch uploaded by brent (license 388)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225003 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue AST-29)
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controls the default disposition of uncommitted transactions.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224446 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@224144 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@224035 f38db490-d61c-443f-a65b-d21fe96a405b
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SWP-151
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https://reviewboard.asterisk.org/r/115/
(closes issue #15820)
Reported by: kshumard
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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: http://svn.digium.com/svn/asterisk/trunk@222398 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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