[asterisk-dev] Fwd: Help editing app_dial for extending L functionality
I've recently introduced in the Asterisk code Now I would like to make some changes in the app_dial But I find one issue: Inside the argument L there is the possibility of streaming a file before timeout (LIMIT_TIMEOUT_FILE), but I cannot figure out in the code where this is being done Any idea about this? Kind regards Manuel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Trying Load a module in asterisk - fails without any message
On Sat, Aug 09, 2014 at 05:46:44PM +0530, Deepak Bhatia wrote: Hello, I had a working application app_mymodule.so which used to work on previous asterisk version but now I am upgraded to 1.8.15-cert5 and now my module doesn't get loaded. Also it doesn't give any error. How exactly are you trying to load the module? If in the asterisk CLI with 'module load app_mymodule.so': what output does it give you? If no output: what is the output of: logger show channels ? Is there some other step required in asterisk 1.8.15-cert5 as compared to previous asterisk version ? Nothing specific I can think of. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Trying Load a module in asterisk - fails without any message
Hi, Thanks In the logger show channels, we get the following output Channel Type Status Configuration --- --- /var/log/asterisk/messages File Enabled- DEBUG NOTICE WARNING ERROR VERBOSE DTMF Console Enabled- NOTICE WARNING ERROR VERBOSE Regards Deepak Bhatia Software Consultant Voxomos Systems Pvt. Limited Mobile: 91 9811196957 C56A/27, Sector 62, NOIDA (NCR), UP, India Skype: toreachdeepak On Sat, Aug 9, 2014 at 6:50 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Aug 09, 2014 at 05:46:44PM +0530, Deepak Bhatia wrote: Hello, I had a working application app_mymodule.so which used to work on previous asterisk version but now I am upgraded to 1.8.15-cert5 and now my module doesn't get loaded. Also it doesn't give any error. How exactly are you trying to load the module? If in the asterisk CLI with 'module load app_mymodule.so': what output does it give you? If no output: what is the output of: logger show channels ? Is there some other step required in asterisk 1.8.15-cert5 as compared to previous asterisk version ? Nothing specific I can think of. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Fwd: Help editing app_dial for extending L functionality
On Sat, Aug 9, 2014 at 7:42 AM, Manuel Camarg sir.lo...@gmail.com wrote:
I've recently introduced in the Asterisk code
Now I would like to make some changes in the app_dial
But I find one issue:
Inside the argument L there is the possibility of streaming a file before
timeout (LIMIT_TIMEOUT_FILE), but I cannot figure out in the code where this
is being done
Any idea about this?
Kind regards
Manuel
The answer to this question will vary greatly depending on your
version of Asterisk. I'm going to use trunk, because that's typically
where new features are developed.
If you are using a different version of Asterisk, your mileage will
definitely vary.
The various L options (including the file to play) are passed to
ast_bridge_timelimit, which is implemented in features.c. From
app_dial:
if (ast_test_flag64(opts, OPT_DURATION_LIMIT)
!ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
if (ast_bridge_timelimit(chan, config,
opt_args[OPT_ARG_DURATION_LIMIT], calldurationlimit))
goto done;
}
The purpose of ast_bridge_timelimit is to set up on the
ast_bridge_config pointer (config) the necessary information to play
back a warning periodically on the channel. It does the following:
* Parses out the time limits/interval periods
* Sets up which files to play based on the variables set on the
channel (or default values)
It does not actually play back the message itself, nor does it create
the structures that will be called to perform the playback. At this
point, we've merely configured the bridge configuration object based
on the parameters passed to app_dial so that the bridging framework
functions can understand what it is we want to do.
Eventually, app_dial will hand the calling channel (chan) and the
dialled channel (peer) to ast_bridge_call. Once upon a time, this
would go into the infamous triple loop of doom in
features.c/channel.c. In Asterisk 12+, this now leads our channels
into the bridging framework. Before that happens, however, we have to
do some work setting up the limits feature on the channel. While we've
parsed out the configuration information into a structure, we haven't
made that information usable. The end goal is to set an
ast_bridge_hook on the channel(s). A bridge hook tells the bridging
core to call a callback function when some criteria is met. In our
case, we want what is known as an interval hook - that is, call some
callback function using some particular time based information. The
act of creating this bridge hook is handled in features.c during the
pre_bridge_setup (called from ast_bridge_call_with_flags):
bridge_config_set_limits(config, call_duration_limits_chan,
call_duration_limits_peer);
if (ast_bridge_features_set_limits(chan_features,
call_duration_limits_chan, 0)) {
abandon_call = 1;
}
if (ast_bridge_features_set_limits(peer_features,
call_duration_limits_peer, 0)) {
abandon_call = 1;
}
Those various function calls will result in the correct interval
ast_bridge_hook being placed on the channels. Internally, this will
call into the constructor callback function for the interval hook that
handles the L option, which is implemented in
bridge_builtin_interval_features.c. Specifically, see the
bridge_builtin_set_limits function, which calls
ast_bridge_interval_hook on the channel's bridge_channel structure.
Once the hooks are on the channel, ast_bridge_call_with_flags will
finish the process of actually putting the channels into the bridging
framework:
/* Put peer into the bridge */
if (ast_bridge_impart(bridge, peer, NULL, peer_features,
AST_BRIDGE_IMPART_CHAN_INDEPENDENT |
AST_BRIDGE_IMPART_INHIBIT_JOIN_COLP)) {
ast_bridge_destroy(bridge, 0);
ast_bridge_features_cleanup(chan_features);
bridge_failed_peer_goto(chan, peer);
return -1;
}
/* Join bridge */
ast_bridge_join(bridge, chan, NULL, chan_features, NULL,
AST_BRIDGE_JOIN_PASS_REFERENCE | AST_BRIDGE_JOIN_INHIBIT_JOIN_COLP);
Without going into the details of the bridging framework, at this
point, you are guaranteed that your interval hook will be called
periodically by the framework.
It's important to note that the L/S options - in Asterisk 12+ - are
built on a system of frameworks. It is completely possible to
implement a different interval hook in a separate module, using the
same frameworks, that would require very few modifications to the rest
of Asterisk. While those frameworks are pretty complex (in fact, this
is probably some of the most complex code in Asterisk - bridging is
hard) - using consistent frameworks helps to improve the stability of
Asterisk tremendously. If you're interested in extending this
functionality, I'd highly recommend using Asterisk 12+.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
Re: [asterisk-dev] [Code Review] 3833: Allow sending voicemail to multiple email addresses
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3833/#review13058 --- trunk/apps/app_voicemail.c https://reviewboard.asterisk.org/r/3833/#comment23499 next appears to be off by one, not generating enough commas. char *next = emailsbuf; fixes the issue for me. - abelbeck On Aug. 8, 2014, 7:15 p.m., Jason Parker wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3833/ --- (Updated Aug. 8, 2014, 7:15 p.m.) Review request for Asterisk Developers and Jacob Barber. Bugs: ASTERISK-24045 https://issues.asterisk.org/jira/browse/ASTERISK-24045 Repository: Asterisk Description --- Allow sending voicemail to multiple email addresses. Should this supersede https://reviewboard.asterisk.org/r/3829/ ? I would think so. Diffs - trunk/configs/samples/voicemail.conf.sample 420313 trunk/apps/app_voicemail.c 420313 Diff: https://reviewboard.asterisk.org/r/3833/diff/ Testing --- Patch in FreePBX distro for ~6 months. Thanks, Jason Parker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3833: Allow sending voicemail to multiple email addresses
On Aug. 9, 2014, 3:42 p.m., abelbeck wrote: trunk/apps/app_voicemail.c, line 5045 https://reviewboard.asterisk.org/r/3833/diff/2/?file=66291#file66291line5045 next appears to be off by one, not generating enough commas. char *next = emailsbuf; fixes the issue for me. Please file an issue on the issue tracker as this patch has been commited. - rmudgett --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3833/#review13058 --- On Aug. 8, 2014, 2:15 p.m., Jason Parker wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3833/ --- (Updated Aug. 8, 2014, 2:15 p.m.) Review request for Asterisk Developers and Jacob Barber. Bugs: ASTERISK-24045 https://issues.asterisk.org/jira/browse/ASTERISK-24045 Repository: Asterisk Description --- Allow sending voicemail to multiple email addresses. Should this supersede https://reviewboard.asterisk.org/r/3829/ ? I would think so. Diffs - trunk/configs/samples/voicemail.conf.sample 420313 trunk/apps/app_voicemail.c 420313 Diff: https://reviewboard.asterisk.org/r/3833/diff/ Testing --- Patch in FreePBX distro for ~6 months. Thanks, Jason Parker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
