lqbal,
I do plan on having alot of users. Two markets I'm trying to get some
volume users from are: residential consumers and business users.
Residential consumers should get basic line services such as their
own DID, voicemail, caller-id, call-waiting, three-way calling, and
basically, all the standard features you get from companies like
Vonage, etc. This particular market base will have a higher volume
than business users.
Business users will get everything residential consumers get, plus
additional features. Features such as, automated attendant, extension-
to-extension calling, company directories, etc.
I guess I would need to have SER and Asterisk work in tandem. Now,
what should be the correct approach in assigning responsibilities to
both SER and Asterisk respectively? Should SER be used strictly as
proxy to Asterisk, may be also registrar, and NAT helper, and then
have Asterisk handle all the calling plans, features, enhanced
services and SER will simply forward everything to Asterisk? Can you
or someone advise as to what would be the more robust/scaleable
architecture to deploy this? Needless to say, it is imperative that I
get proper CDR from either one or both systems in order for me to
properly bill our users. I don't know which of the two platforms has
a more robust/customizable call logging facility.
I took the liberty of cross-posting to the Asterisk list in order to
get some of their feedback as well.
Thanks,
Waldo
On Aug 23, 2005, at 6:49 AM, Iqbal wrote:
Um..no actually I am saying you could combine both,
but that will
only help if you have alot of users. I guess you could direct calls
to a particular sip client, ut normally when ser and asterisk work
in tandem, all calls from SER hit one section of sip.conf, and
hence can only be pointed to one context, you can get around this
by including contexts from this default one, which is what I do,
based upon a mysql lookup, but then you will have problems in call
pickup, because all pickup is not context based, again there is a
solution to this, if you look at bristuff patch for asterisk.
If you dont have many users stick with ust asterisk, if you want to
scale you may need to kludge something with ser and asterisk, and
this might be easy or hard depending on exacly what you require,
and call scenarios.
Iqbal
Waldo Rubinstein wrote:
The way I manage this in Asterisk is every SIP UA
has a unique
login but in different contexts. I suppose that if SER directs a
call to Asterisk to the specific SIP client, Asterisk will
recognize it belongs to a different context. The question is, I
don't know if SER knows about multiple contexts under the premise
of the Asterisk world.
Also, I get the feeling you are pretty much telling me to stick
to Asterisk :) Is that so?
Thanks,
Waldo
On Aug 22, 2005, at 3:26 PM, Iqbal wrote:
Hi
If you are already using multiple contexts within asterisk, then
your already half way there, the problem is if you stick in SER,
bcause then your phones are not registered in asterisk, hence
all fall into the same context in sip.conf, which means they
all will hit one context in extensions.conf, hence you should
look into that.
I am not sure if you can do the 101/102 extension thing in
asterisk, since aliases will be bound to a contact, whereas in
asterisk the context is also part of the dialing plan.
DID can be done, as can forking and directing to voicemail on no
answer.
Iqbal
Waldo Rubinstein wrote:
Hello,
I'm still trying to learn more about SER. I've been using
Asterisk to manage virtual PBX services for different
companies by using multiple contexts within Asterisk. However,
since I only use Asterisk with SIP UAs and to communicate with
ITSPs, I don't have the need to have all the fancy features
Asterisk offers, plus I have the additional advantage of
having the built-in NAT support in SER.
The question I have is if someone can point me to the right
place where I can see some sample configs that do more or less
the things I need or if someone would be willing to share some
of those configs on the list.
Basically, I need to have the ability to manage any number of
virtual PBX services where each virtual entity can manage
their own extension numbering, DIDs, outbound CLID. I would
probably just continue using Asterisk for voicemail services
since I get the feeling that SEMS is still "unstable" to go
into production, even for voicemail services only (unless told
otherwise).
The idea is that even if two companies define two extensions
101 and two extensions 102, when each company calls ext 101 or
102 internally, the call will be maintained within that
company and would not cross over to the other company. Also,
having features like one main number (DID) that could ring in
multiple extensions simultaneously (forking) and if no one
answers, leave a message in the general mailbox, or that each
extension could have, additionally, a DID so that they can be
reached directly from the outside.
Any help would be greatly appreciated.
Thanks,
Waldo
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