[Asterisk-Users] SER and Asterisk

[Yair Hakak]

On 10/19/05, Yair Hakak <yhakak at gmail.com> wrote:
>
> i do it this way because i want all the dialplan logic and CDR having to
> do with PSTN in asterisk, not SER.
> so, calls from the outside are adressed to sipaddress at myserver:5070 and
> hit asterisk. asterisk either sends them along to 5060, or handles them
> internally (IVR, voicemail, etc) based on the dialplan.
> clients on the inside are registered to the SER at 5060 and the SER
> automatically forwards them to asterisk. if they are PSTN asterisk serves as
> PSTN gateway, if they are internal, asterisk native bridges and drops out,
> but still keeps the CDR (i have full SIP addresses in my dial statements
> instead of asterisk SIP peers)
> the reason i do this is i found that if the endpoints are scattered on the
> internet, SER+rtpproxy is much more stable than asterisk as a SIP server
> (asterisk kept dropping endpoints). This way SER serves as a completely
> "dumb" SIP server, and just sends everything along. there is a minimal
> increase in overhead (i could handle internal calls just with SER) but it's
> worth it to have all the dialplan logic and CDR's in one place.
>  also, obviously, if i use an IAX provider for outgoing, asterisk has to
> be in the middle.
>  i agree though, it makes more sense to have SER on 5060 and asterisk
> somewhere else.
>  hope i'm making some sense, please point out if i'm doing something
> really stupid.
>
> -yair
>  On 10/19/05, trixter aka Bret McDanel <trixter at 0xdecafbad.com> wrote:
>
> > On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote:
> > > hello,
> > > trace the SIP packets and see if they are actually addressed to 5062.
> > > if you post the ngrep or ethereal dump we'll see whats actually going
> > > on. I do this with SER on 5060 and asterisk on 5070 and there are no
> > > problems - my extensions point to 5060 and my DID's point to 5070 so
> > > asterisk serves as the gateway to the PSTN.
> > >
> > > -yair
> > >
> > >
> > also look for dns packets and see if htey are pulling the server info.
> > Some sip clients look for specific server type dns records to see where
> > they should go.
> >
> > 5060 is the default, wouldnt it make more sense to have the default port
> > be what you want the devices to goto and have that proxy to the device
> > you dont want direct connectivity to? Or am I missing something in that
> >
> > >
> > --
> > Trixter http://www.0xdecafbad.com Bret McDanel
> > UK +44 870 340 4605 Germany +49 801 777 555 3402
> > US +1 360 207 0479 or +1 516 687 5200
> > FreeWorldDialup: 635378
> >
> >
> > -----BEGIN PGP SIGNATURE-----
> > Version: GnuPG v1.4.1 (GNU/Linux)
> >
> > iD8DBQBDVg5f+1olxlzQw5cRAhl5AJ91lwjqMb2EPcDSXH69dOELBOq0IQCgvr8m
> > 4NqQAGLmWLokUXjl7Bi7SbI=
> > =thAz
> > -----END PGP SIGNATURE-----
> >
> >
> >
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051019/2602326b/attachment.htm