One potential issue tho is that I want to be able to click-to-dial on just extensions as well as external numbers. So if I click ext-300 or x300 it dials 300 with no prefix, but if I click 123-555-1212, it uses a 9 prefix.
It'd be cool if we could provide a pseudo-dialplan for it ...
... or something. So the above would not prefix anything matching 3XX or 6XX, but would prefix everything else with a 9.
A global callee prefix would be useful, regardless.
And yeah, I'm already handling my local extension calling with a custom regex that just adds my local extension pattern(s) to my already modified "US calling plan" regex.
I've gotten things working for outside calls and local extensions by temporarily modifying my main Outbound Route in Asterisk to not require the 9 for an outside line. Right now this is a test setup, but I have to deliver it to a customer who is going to want the 9 prefix for external calls, something to do with their billing software that processes the Asterisk reports.
What might work is if you added a 'replace' regex. So anything which matches the main regex, you add the global callee prefix, then run it thru the replace regex (if specified).
Then I could have a replace regex that does s/9(\d\d\d)/$1/, i.e. removes the 9 from 3 digit numbers.
Not a huge deal, but something to think about.
There is a new version of the Asterisk PBX Interation Zimlet available on sourcforge.
Basically there are 2 new global config options 'calleePrefix' and 'numberCleanRegExp'. Please check Changelog for more details.
First off thanks for an amazing zimlet! I have used it every day since installed and must say its rock solid as far as I can tell.
Today I embarked on getting my colleagues to also use it - but I encountered a problem;
We use a pbx from a company named Diptel (link) - It is based on asterisk 1.4. It is the one I have configured for my account now, so it is working; but it takes complete control of the asterisk dialplan and it places every user in a unique context they need to use in order to get the right MSN number when calling pstn.
For now I have created a common user that has our main switchboard number as MSN, and use that context, but that means our customers will see the main switchboard number calling, not the agents personal extension - it could create some confusion if not answered and the customer dials back later on hitting the switchboard that does not know who called who
Would it be possible to allow the context to be configureable on a per-user basis? And, for scalability in the future - maybe the server/user/pass fields also since I can foresee scenarios where I might place some of my users on a separate pbx, e.g. a branch office or something...
Well.. just my 2c.. thanks again for this great piece of code
A user configurable context is rather easy to implement. I am just a bit skeptical because of security reasons. I could let the global option and if present it would always override the user preference. So its up to the admin to decide. But it think its not worth to implement user configurable AMI user/pass. I would prefer same AMI user/pass for all asterisk instances than user editable one's(that are even readable when using a JSdebugger in the Browser).
You don't have multiple zimbra hosts, dou you? There would be the possibilty to user host based options instead of global one's. But of course these refer to the zimbra host. So if there is a 1:1 relation of zimbra hosts to asterisk instances it might be a solution.
I've created a 0.63 beta with a user configurable Context. You must let the astDialContext empty in global config, otherwise usersetting is not working.
Let me know if it works.
Are there any plans to integrate a view to the Astrisk VoiceMails ? I have seen the comcast Screenshot here:
Zimbra voice mail image - Images: Comcast gets Zimbra zing - CNET News
and read about the Digium/asterisk intergartion here:
Asterisk Voicemail Zimlet - Zimbra :: Wiki
But I dont know how this is related to your integration efforts....
Thanks for any help or advice,
Now there are currently no plans for integration voicemails. There are 2 main rasons:
1) I don't know any way to get the voicemail streams using AMI. (Correct me if i am wrong!) Which means you need to install some funny scripts on the Asterisk Server to get the Voicemails. Maybe there would be someway when unsing AJAM, but that will break compatibility with asterisk < 1.4
2) You can easily configure astrerisk to send the voicemails as Email to your Zimbra Account.
For now I think its not worth all the work. But its not forbidden to contribute development :-)
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