25 Oct
2005
25 Oct
'05
7:05 p.m.
Okay, that sounds similar to my setup, all my sip to sip are within ser,
only voicemail etc go through asterisk, although I dont put Sip to pstn
through asterisk either...so we differ there.
In the latest version (beta) of asterisk you can get alot of sip header
info per call in the dial plan, I havent installed it, but its one of
the major features of it. The context problem is a problem, I use
asterisk to do virtual PBX solution, however to do this I simlply fire a
mysql query based upon the callid, and select the context entry, and
then goto the context that is matched. This I will replace when beta
becomes stable.
If you havent see if you can pull these fields from the latest asterisk,
if you cannot you could do it via mysql in the dialplan. I think it
makes better sense to just have one entry into sip.conf, and then direct
into one main context and then include/goto the others based upon
that...but I have been wrong in the past :-)
Iqbal
Barry Flanagan wrote:
Iqbal wrote:
Wouldnt you then be authenticating twice,
openser, and asteriks, or
do you want to drop the openser auth.
Just curious as to why you want to auth in asterisk
Because I want to have asterisk know the context, callerid and the
accounting code so that the user can be billed properly and put into
the correct context.
I don't actually want the user to have to input their credential twice
- I am assuming that if the realm, username and password are the same
that this will allow authentication to take place. Is this incorrect?
Essenstially, I want users to be registered on openser, and that all
sip to sip local calls happen within openser. Calls to and from the
PSTN, and voicemail, etc. I want to go to/from asterisk.
-Barry
Iqbal
Ray Van Dolson wrote:
> Barry, I'm curious if you'll find a solution for this.
>
> Currently I'm using SER with Asterisk and I am just straight proxying
> everything through to Asterisk. Registrations in this case work
> just fine,
> but the voice path seems to always include Asterisk with this setup :(
>
> What happens when Asterisk sends an Unauthorized (4xx) response
> after one of
> your ATA's sends an INVITE and it gets proxied through to Asterisk?
> Maybe
> you could get OpenSER to trigger on this and then the next REGISTER it
> receives from the same ATA gets proxied on to Asterisk instead of being
> processed locally.
>
> Just thinking out loud.
>
> Ray
>
> On Tue, Oct 25, 2005 at 05:34:24PM +0100, Barry Flanagan wrote:
>
>
>> Hi,
>>
>> I want to use openser as proxy/registrar with Asterisk. Users will
>> exist both in openser's subscriber database and in asterisk's sip
>> friends.
>>
>> I have this kind of working, but have a few questions.
>>
>> Currently openser authenticates users fine, and will route to * for
>> pstn and asterisk will accept the call if I have openser set up in
>> * sip.conf as a proxy.
>>
>> I would rather, however, that the user gets authenticated as
>> themselves on asterisk, rather than the call happening as the proxy
>> user. without openser set up in asterisk, I get continual 407 Proxy
>> Authentication required.
>>
>> Additionally, it appears that the call is still routing through
>> openser, rather than being directly between the client and asterisk.
>>
>> Questions:
>>
>> 1. How can I set it up so that when a call gets sent to * that
>> authentication takes place directly between asterisk and the client
>>
>> 2. How should I be sending the call from openser so that the rest
>> of the call is directly between asterisk and the client?
>>
>> Any working configs would be welcome - these seem to be hard to
>> come by.
>>
>> Thanks.
>>
>
>
>
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