{5} Assigned, Active Tickets by Owner (Full Description) (19 matches)

2.1beta The callback module would it be posible to have some logic for the callback number

as in if you have not entered a number and the caller id is missing then ask the caller for a number.

This module supports Festival / Flite and Cepstral text to speech engines (autodetect and let the user choose which engine he wants to use).

A TTS record can be used as a destination and can be linked to another destination after being played.

The installer, if it can't connect to the database with the username/pass supplied in amportal.conf, should ask if the user wants to change them (and then prompt for new ones) or create a new database.

If they want to create a new db, it should prompt for an admin user and pass (that has permissions to create the database), then run CREATE DATABASE asterisk, and finally, import the install.sql file into it.

It should probably also warn if they're using the default db password that they should change it.

As part of the Dial Plans etc   For example here in New
Zealand the typical Dial Prefix (to get an outside
line) is 1, the emergency number is 111 and you dial 0
for national calls, 00 for international calls etc
(this is a similar standard to in europe).

It would be nice to have under general settings a drop
down box that you can pre-configure the country
specific dial plan, then customise it where necessary.

To add on to the request the abiltity to group
extensions into particular call plans.  Ie this group
of extenstions can make national calls etc.

An AMP user should have one web page to access all the
AMP information relevant to that user. Here is an
example of some of the functions that the user could
view and change.

E-mail address that voicemail is sent to
Users extension (read only)
Current voicemails (play, delete, etc)
Users calls logs (calls made to/from users extension)
Quick links to the last 10 incoming / outgoing calls
with click to re-dial
Personal speed dial list
Extension call routing/personal auto-attendant

Most of there feature are self explanatory

-Extension call routing/personal auto-attendant
This would allow a user to pick what happens when their
extension is dialed. They could select or deselect DND
or they could add a call rule.

For example if the time is between 9am and 5pm ring
extension then go to voicemail. If the time is 5pm ;
10pm ring cell phone 3 times then go to voicemail.
10pm-9am ring all three on-call support number and then
go to voicemail.

Each rule would have a time that it is active and
steps. Each step would be done after or in parallel
with the previous step. So each step would have thee
piece if info.
1.	what to do -  call a number, go to voicemail, go to
an     auto-attendant
2.	what number, aa to call
3.	how many rings
4.	if this is a sequential step or is done in parallel
with the last step

This is a lot of features in one. I think the Extension
call routing/personal auto-attendant is the most
important part if there is only time to implement part
of this.

Need a way to detect quality of VoIP providers (or
rather, *'s connection to them) and block use if
quality is too low.

This will be implemented with a flag in the * db, like
BLOCKTRUNK/{trunknum} 1. Dialout macro will look at
this flag, and skip dialing that trunk (go on to the
next priority trunk in the route) if it exists. (thanks
to Ferrari_ for suggestions) (anyone have a better
alternative to 'BLOCKTRUNK'?)

Metrics of the link can be determined with something
simple like ping/jitter, and a threshold.. optimally I
suppose, would be a loopback and some way of detecting
the quality returned.. this would be much more complex
to implement, and hard logistically to make it work
anywhere in the world (availability of loopback,
connectivity of voip provider, endpoint (PSTN, another
* box, some other voip system?)).

should there be an interface on the trunk config screen? 

[ ] Monitor connection quality
    Don't use if ping time is over ___ms [AND/OR]
varies by more than ___ms. 


There would just be a script that ran every minute/5
minutes/whatever that would run the metric tests, and
block connections as approiate. Using this method,
current calls using the channel would not be
interrupted, just new calls.

Looking for any additional feedback before I implement
this feature (hopefully sometime next week.. probably
inversely proportional in time to the amount my VoIP
providers piss me off..)

 Hi! 
 
ARI is very usefull, but it lucks options of complex
search with sorting. 
I'm trying to add links from AMP reports to ARI, to be
able to make a complex search with reports menu, and
then just by click to go specific record in ARI. 
 
In DB there's a uniqueid field, and I can find a
recording from url like: 
http://192.168.0.37/recordings/index.php?s=callmonitor&q=asterisk-13817-1131353495.213&start=0&btnS=Search

http://.....uniqueid..... 
So it's enouth to display a link in reports.php with
uniqueid to be able to access a recording from
reports.php. 
 
So if I add 'uniqueid' in call-log.php in the line: 
$FG_COL_QUERY='calldate, channel, src, clid, dst,
disposition, duration, uniqueid'; 
 
and add a line: 
$FG_TABLE_COL[]=array ("Recording", "uniqueid", "6%",
"center", "", "30"); 
 
It displays 'uniqueid' for every call in reports. 
 
If it will display it in the form of URL like in my
example: 
http://192.168.0.37/recordings/index.php?s=callmonitor&q=asterisk-13817-1131353495.213&start=0&btnS=Search

 
It will be possible to go directly to record for
speciffic call from reports. 
 
I'm not to strong in PHP :( 
Can somebody help me please to add this feature to AMP? 
I think it's nice to be able to see recordings from
reports. 

When defining some custom devices, freepbx
automatically associate a hint for the extension. This
causes some trouble when we use these custom device to
reach some remote phones like mobile phones or remote
collaborators with a local user extension.

Exemple:
I define a device extension for a mobile phone, in the
GUI (admin/setup/device/custom), I associate the device
to a user extension (let say 804) and I define de dial
extension in the dial field: "IAX2/TRUNKNAME/PHONENUMBER".

I then automatically have a hint wich maps the given
dial  extension in the [ext-local] context
(extensions_additional)

exten => 804,1,Macro(exten-vm,novm,804)
exten => 804,hint,IAX2/TRUNKNAME/PHONENUMBER

But in this particular case, it is really cumbersome,
and does not work. Extension is always unavailable. I
would prefer to have something like that:

exten => 804,1,Macro(exten-vm,novm,804)
exten => 804,hint,IAX2/TRUNKNAME

to associate the availability of the user to the trunk
availability itself. Modifying it by hand works great
actually. But at this stage there is no way to override
the default config from the GUI (which is restored
after every reloadconfig). That would be a very nice
feature to have it permanently ? We could imagine a
field in the device configuration page to allow to
specify the hint for every device and default to the
present behaviour.

Another interesting possibility to avoid the rewrite of
the custom modifications is to put the hint config for
device extensions in the SQL DB.

Thanks for your comments

I would like to see a module that allows you to define certain extensions or codes (e.g. *30) so that they will do the following:

- Play ringing tone to the caller

- Download an audio file from a specific fixed location OR extract the location URL from a "podcast" XML file and download that audio file (but, in the case of a podcast file, don't download the file again if we already have it from a previous call and it hasn't been updated - just play the one we previously downloaded).

- When a new file is completely downloaded, call sox to convert the file to a format asterisk is capable of playing (if necessary)

- answer the call (stop the ringing tone)

- play the file to the caller

- hangup

- Delete the file, unless it's a podcast file in which case save it (until a newer one is available to overwrite it) because someone else may want to hear it.

When I first started playing with Asterisk, I did it this way (this does NOT work anymore, because of some change in how Asterisk 2.7 handles shell scripts):

In extensions_custom.conf:

exten => *30,1,ringing
exten => *30,2,System(/var/lib/asterisk/batch/getmsnbc.sh)
exten => *30,3,Wait(1)
exten => *30,4,Answer
exten => *30,5,MP3Player(/tmp/msnbc.mp3)
exten => *30,6,Hangup

Then in getmsnbc.sh (don't laugh too hard, it was my first Linux script and it took me hours to write, but in the end it worked, although it probably does some things that don't need to be done):

if [ -a /var/lib/asterisk/batch/msnbc.xml ]
then
mv -f /var/lib/asterisk/batch/msnbc.xml
/var/lib/asterisk/batch/msnbc2.xml
fi
/usr/bin/curl -s
http://podcast.msnbc.com/audio/podcast/MSNBC-Headlines.xml
> /var/lib/asterisk/batch/msnbc.xml
if cmp -s /var/lib/asterisk/batch/msnbc.xml
/var/lib/asterisk/batch/msnbc2.xml
then
exit 0
else
grep -m 1  "http://podcast.msnbc.com/audio/podcast/vh"
/var/lib/asterisk/batch/msnbc.xml | sed 's/^[
\t]*<link>/\/usr\/bin\/curl -s /;s/<\/link>/ >
\/tmp\/msnbc.mp3\nexit 0\n/' >
/var/lib/asterisk/batch/msnbcurl.sh
chmod 744 /var/lib/asterisk/batch/msnbcurl.sh
/var/lib/asterisk/batch/msnbcurl.sh
fi
exit 0

Yes, this really did write a second shell script, then jumped into it. Sorry but as I say, this was my first attempt at trying to make a shell script. In Asterisk@Home 2.4 it would execute both scripts, and only then return to exten => *30,3,Wait(1) But in 2.7 it seems to fire off the shell script and continue blindly on, without waiting for the script to finish. The scripts themselves still work, but A@H doesn't wait for them to finish.

So, in the module it should ask for these things:

The extension number you want to assign this to (could be a * code also)

The full path and filename of the audio file or podcast (xml format) file.

Select ("radio button") whether it's a direct link to an audio file or a podcast XML file (if this can't be determined automatically - note that podcasts do not always use a file with an XML extension, so you can't necessarily go by extension)

And, for XML-format files, the XML tag containing the path to the actual file. This could be a little tricky because, for example, in this case you're looking for the tag "<enclosure url=" but the data you want is INSIDE the tag, such as this:

<enclosure
url="http://podcast.msnbc.com/audio/podcast/vh-03-25-2006-161811.mp3"
length="108239" type="audio/mpeg"/>

Point is, you need some way to explicitly declare where to look in the file for the URL of the file to download and play. I am led to believe "<enclosure url=" is a normal tag for a podcast BUT I can imagine that in some other cases the desired URL would be bracketed by specific tags (not part of the tag itself). Also there may be times when you might want to extract a URL from a regular HTML page.

Bonus points for allowing playback of formats other than MP3. TRIPLE bonus points if you manage to play the dreaded WMA format and its ilk.

Note we are NOT talking about playing STREAMING audio here, though that could certainly be another option. These are for things like one or two minute newscasts, weather reports or whatever. Just an idea (since it's much easier to pick up a phone and punch in a few digits than to go to a computer, fire up a web browser, find what you want to listen to, and try to launch it).

Hi guys, here is a very simple module that can make Snom 360 users happier ;)

It gives you direct access to the phonebook module, to the company directory and to the feature codes of freePBX using the minibrowser feature of firmware version >= 6.x It allow you to direct dial each entry.

It's still in beta stage and it's never been tested with a real Snom 360 because I don't have one of them available at the moment.

This is meant to be a starting point to make freePBX taking advantage of extra features of various VoIP phones (AFAIK Cisco is also supporting some kind of XML integration and quite every phone model supports an internal phonebook)

Maybe this module is somehow related with DevicesTakeTwo discussion

Have fun