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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.22-rc4@212958 ↵v1.4.22-rc4kpfleming3-9/+8
f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-13document dahdichanname option in doc/asterisk-conf.txtkpfleming1-1/+7
make chan_dahdi read its configuration from zapata.conf if dahdichanname has been set to 'no' git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@137527 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-13Update docs to reflect the change to cdr_tdsseanbright1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@137405 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-22The most common question on the #asterisk iRC channel and on mailing listsoej1-0/+126
seems to be in regards to an error message when retransmit fails. This is frequently misunderstood as a failure of Asterisk, not a failure of the network to reach the other party. This document tries to assist the Asterisk user in sorting out these issues by explaining the logic and pointing at some possible causes. Hopefully, we will get other questions now :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@132645 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11a whole pile of Zaptel/DAHDI compatibility work, with lots more to come... ↵kpfleming1-5/+5
this tree is not yet ready for users to be easily upgrading or switching, but it needs to be :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@130298 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11new installations should be using DAHDI instead of Zaptel, so the sample ↵kpfleming3-7/+7
config file is now chan_dahdi.conf instead of zapata.conf also, convert remaining references to zapata.conf in various places git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@130042 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Update documentation to have the correct option namemmichelson1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@129208 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Backport TCP-related timeouts to IMAP voicemail in 1.4mmichelson1-0/+4
since it should solve bugs people are experiencing. Specifically, there are times where communication with the IMAP server causes system calls to block forever. If this should happen when querying the mailbox so that chan_sip's do_monitor thread can send MWI to a phone, it means that SIP calls cannot be processed any more. The timeout options are outlined in doc/imapstorage.txt. Defaults for the timeouts are sixty seconds. (closes issue #12987) Reported by: mthomasslo git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@129158 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-20Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905jeffg2-41/+57
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@124372 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-18Add support for saying numbers in Hebrew.tilghman1-0/+0
(closes issue #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008 uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods uploaded by greenfieldtech (with signficant changes to the spreadsheet by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@123769 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-11add instructions for logging gdb output via set logging onjpeeler1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@121804 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-05It turns out that searching on the forwarding station isn't very useful forrussell1-18/+130
most people, so pull in the changes that allow searching for SMDI messages based on other components of the SMDI message. Also, update the SMDI documentation. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@120671 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-23Add information on using the Asterisk console, including tab command linetilghman2-0/+34
completion. (Closes issue #12681) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@118052 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28Update debugging text, since Valgrind eliminated the --log-file-exactly option.tilghman1-1/+8
(Closes issue #12320) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@111605 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Update documentation for pgsql ODBC voicemail.qwell1-24/+50
(closes issue #12186) Reported by: jsmith Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license 15) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@107826 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05document var_metric so no bugreports will come in when it's actually a ↵mvanbaak1-0/+14
configuration issue. (issue #12151) Reported and patched by: caio1982 1.4 patch by me git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@106178 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-15Final round of changes for configure script logic for IMAPmmichelson1-1/+4
Now if a directory is specified, then we will search that directory for a source installation of the IMAP toolkit. If none is found, then we will use that directory as the basis for detecting a package installation of the IMAP c-client. If that check fails, then configure will fail. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@103722 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-15I apparently misunderstood one of the requirements of this configure change.mmichelson1-4/+2
Now, if a source directory is specified with the --with-imap option, and a valid source installation is not detected there, then configure will fail and will not check for a package installation. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@103709 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-14Make a small clarification in the documentationmmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@103703 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-14Update documentation regarding configuration of IMAPmmichelson1-8/+18
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@103701 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Backport the ability to set the ToS bits on Linux when not running as root.russell1-0/+7
Normally, we would not backport features into 1.4, but, I was convinced by the justification supplied by the supplier of this patch. He pointed out that this patch removes a requirement for running as root, thus reducing the potential impacts of security issues. (closes issue #11742) Reported by: paravoid Patches: libcap.diff uploaded by paravoid (license 200) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@98265 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20closes issue #11324; break statements missing in switch cases.murf1-10/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89450 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19Clarify documentation a bit, include that a frame has to pass through the ↵file1-1/+1
core in order for the Local channel optimization to happen. (closes issue #11246) Reported by: jon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89416 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Typotilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89103 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Add some notes on using valgrindtilghman1-0/+19
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88116 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-12Change Digium addresstilghman2-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85523 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-24Oops. Removed the unworkable workaround. This note should never have been ↵tilghman1-3/+1
in the release. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@83653 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-14Fixing a typo in the coding guidelinesmmichelson1-1/+1
(closes issue #10717, reported and patched by leedm777) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82376 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-22Merge changes from team/russell/iax_refcount.russell1-1/+1
This set of changes fixes problems with the handling of iax2_user and iax2_peer objects. It was very possible for a thread to still hold a reference to one of these objects while a reload operation tries to delete them. The fix here is to ensure that all references to these objects are tracked so that they can't go away while still in use. To accomplish this, I used the astobj2 reference counted object model. This code has been in one of Luigi Rizzo's branches for a long time and was primarily developed by one of his students, Marta Carbone. I wanted to go ahead and bring this in to 1.4 because there are other problems similar to the ones fixed by these changes, so we might as well go ahead and use the new astobj if we're going to go through all of the work necessary to fix the problems. As a nice side benefit of these changes, peer and user handling got more efficient. Using astobj2 lets us not hold the container lock for peers or users nearly as long while iterating. Also, by changing a define at the top of chan_iax2.c, the objects will be distributed in a hash table, drastically increasing lookup speed in these containers, which will have a very big impact on systems that have a large number of users or peers. The use of the hash table will be made the default in trunk. It is not the default in 1.4 because it changes the behavior slightly. Previously, since peers and users were stored in memory in the same order they were specified in the configuration file, you could influence peer and user matching order based on the order they are specified in the configuration. The hash table does not guarantee any order in the container, so this behavior will be going away. It just means that you have to be a little more careful ensuring that peers and users are matched explicitly and not forcing chan_iax2 to have to guess which user is the right one based on secret, host, and access list settings, instead of simply using the username. If you have any questions, feel free to ask on the asterisk-dev list. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80362 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Fix mogs email address.qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@78646 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26change protocol for downloads as wellkpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77429 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26use new canonical name for download serverkpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11further reversion of previously applied floating point stuff for expr2murf1-14/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@74628 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-02support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes ↵murf1-7/+14
bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@72933 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-25Fix a typo in the Asterisk mib. (issue #10048, Matti)russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@71519 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31Issue 9850 - update preferred command line syntaxtilghman1-4/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66821 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-29Update datastores documentation. (issue #9801 reported by mnicholson)file1-9/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66398 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-16Making these documentation changes in the 1.4 branch upset various people, sorussell53-5284/+2732
these chanes will only be done in the trunk. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58955 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Refashion dump command to match common syntax and update the resulting ↵tilghman1-0/+87
appdocs TeX file git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58946 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15add a link to the rubber homepagerussell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58941 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Add Asterisk version information to the generated PDFrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58937 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Merge changes from svn/asterisk/team/russell/LaTeX_docs.russell53-2732/+5195
* Convert most of the doc directory into a single LaTeX formatted document so that we can generate a PDF, HTML, or other formats from this information. * Add a CLI command to dump the application documentation into LaTeX format which will only be include if the configure script is run with --enable-dev-mode. * The PDF turned out to be close to 1 MB, so it is not included. However, you can simply run "make asterisk.pdf" to generate it yourself. We may include it in release tarballs or have automatically generated ones on the web site, but that has yet to be decided. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58931 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-14Add a more basic example setup to the examples sectionrussell2-30/+69
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58902 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-14Merged revisions 58896 via svnmerge from russell1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines Add a note to the security file that the Asterisk CLI and log files may contain sensitive information, and that people should keep this in mind. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58897 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-09Merge some updates to the SLA documentation. I plan to keep working on thisrussell3-225/+339
to explain all of the expected behavior with call handling, configuration details for specific phones, and other things. However, I got tired of doing it in plain text, so I switched to using LaTeX. I have included the PDF version. I haven't been able to get a nice looking plain text version out of it yet, but I'm not terribly concerned since this is supposed to be more of the manual, while the plain text sample configuration file is the reference. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58638 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merge changes from svn/asterisk/team/russell/sla_updatesrussell1-5/+137
* Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57364 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-011. Corrected a typo for www.etsi.org. Thank Patrick.transnexus1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57263 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28minor tweaks to the sla docsrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22Merge changes from team/russell/sla_updates.russell1-11/+11
This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56277 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22Update OSP documentation for v1.4.transnexus1-418/+376
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56169 f38db490-d61c-443f-a65b-d21fe96a405b