[OpenSIPS-Users] New: Closeddial module

[Jeff Pyle]

Brett,

Would functionality like this make sense in a proxy?  I¹m thinking of all
the things that wouldn¹t work.  Call transfer (REFER) comes to the front of
my mind.


- Jeff



On 2/9/09 1:46 PM, "Brett Nemeroff" <brett at nemeroff.com> wrote:

> I think the basic idea is to provide PBX routing like functionality..
> Ultimately, it would have:
> 1. Extension Mapped to Login (login used for register)
> 2. Extensions within the same group are dialable
> 3. Can't dial extension to extension if groups aren't the same
> 4. Naturally, extension numbers can be duplicated as long as group id differs
> 5. Full phone number (DID) mapped to Login
> 6. DIDs *can* be dialed if groups differ *with module parameter flag* (ie
> allow_intergroup_did_dial=1), in other words, it should be an option to hide
> the DID so that direct dialing between customers isn't allowed and instead
> must traverse LCR.
> 
> I kind of imagine that upon receiving an INVITE, we'd lookup the group id
> based on an avp. Then pass that to a new fancy lookup() function ie:
> lookup($avp(s:groupid))  which would return the registered URI for that
> did/exten. I do think it's necessary to distinguish if a DID or an Extension
> is being dialed for many reasons:
> 1. Caller ID Display name may be different for internal calls (will transmit
> extension number for example, and station name)
> 2. E911 ANI may be different for outbound calls
> 3. Transmitted ANI for regular outbound calls may want to mask station's
> callerid
> 
> Of course, a lot of this can be done with aliases, but I think this is a more
> sophisticated approach that would provide for some real usability that would
> result in configuration files that are much more readable.
> 
> That's my $0.02. I like the direction you've gone with this. Hope you don't
> mind my feedback. I very much appreciate your contributions!!
> 
> -Brett
> 
> On Sat, Feb 7, 2009 at 7:51 PM, Sergio Gutierrez <saguti at gmail.com> wrote:
>> 
>> Hi Brett.
>> 
>> Thanks for your comment; the idea is interesting. I will have it present for
>> a next release of module.
>> 
>> The approach of integrating into register is really interesting; anyway, I
>> hope you find useful the module in its current stage.
>> 
>> Thanks again.
>> 
>> Regards.
>> 
>> Sergio
>> 
>> 
>> On Sat, Feb 7, 2009 at 8:21 PM, Brett Nemeroff <brett at nemeroff.com> wrote:
>>> This is interesting, but I wonder...
>>> 
>>> Seems like it would be more useful if this was integrated also into the
>>> register function so that dynamic clients can be grouped.
>>> 
>>> Said another way, seems like "new_uri" doesn't really make sense. It should
>>> point either a new_uri or to an already registered contact. If it was
>>> integrated into register, you'd put the user into a group when they
>>> register, then the lookup function would also need to pass the group.
>>> 
>>> Maybe I'm thinking about this wrong.. Good idea tho. :)
>>> -Brett
>>> 
>>> 
>>> On Sat, Feb 7, 2009 at 6:04 PM, Sergio Gutierrez <saguti at gmail.com> wrote:
>>>> Hello to all developers and users.
>>>> 
>>>> I just have commited a new module to OpenSIPS which is called closeddial.
>>>> 
>>>> This module is intended to offer a functionality similar to Centrex to
>>>> OpenSIPS, allowing to define groups of closed dialing, using abbreviated
>>>> codes.
>>>> 
>>>> Closeddial uses a database table, where the relationship between users,
>>>> their abbreviated dialing codes and their grouping through a particular
>>>> attribute is stored.
>>>> 
>>>> An example which illustrates the idea behind module:
>>>> 
>>>> Supposing that the following is the content of closeddial table on database
>>>> 
>>>> User            closeddial_code          group_id           new_uri
>>>> 135             00                              companyA
>>>> sip:123 at proxy1.com <mailto:sip%3A123 at proxy1.com>
>>>> 357             01                              companyA
>>>> sip:357 at proxy1.com <mailto:sip%3A357 at proxy1.com>
>>>> 579             02                              companyA
>>>> sip:579 at proxy1.com <mailto:sip%3A579 at proxy1.com>
>>>> 024             00                              companyB
>>>> sip:024 at proxy1.com <mailto:sip%3A024 at proxy1.com>
>>>> 246             01                              companyB
>>>> sip:246 at proxy1.com <mailto:sip%3A246 at proxy1.com>
>>>> 468             02                              companyB
>>>> sip:468 at proxy1.com <mailto:sip%3A468 at proxy1.com>
>>>> 
>>>> Users defined within group companyA can use abbreviated codes to dial to
>>>> others users, instead of using full username; their abbreviated codes will
>>>> not collide with codes defined for group companyB, because group_id is
>>>> determined by using from username, before looking the uri to rewrite.
>>>> 
>>>> group_id value can be left to be determined by querying database, or can be
>>>> passed from script, in case it be determined from other mechanism (for
>>>> example, an avp loaded at register time).
>>>> 
>>>> I hope this module be useful in your deployments of opensips.
>>>> 
>>>> Feel free to send me any doubts or feedback about module, through users
>>>> lists, or directly to my mail.
>>>> 
>>>> Best regards.

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