[Asterisk-Users] SER & Asterisk

[Jess Magnaye]

I tried the following setups. Although there is a minor "ringback" issue
that I haven't found any solution yet.

1.) ATA->CiscoNAT->Asterisk->SER+RTPProxy->Cisco2600

2.) ATA->CiscoNAT->SER+RTPProxy1->Asterisk->SER+RTPProxy2->Cisco2600

I cannot remember if * can directly connect to Cisco2600. I know I had
problems initially with it, that's why I installed the SER, and since now
I'm focusing to solve the ringback issue, I didn't have time to take out SER
out of my equation.



----- Original Message ----- 
From: <asterisk at geek.be>
To: <asterisk-users at lists.digium.com>
Sent: Friday, January 16, 2004 7:33 PM
Subject: Re: [Asterisk-Users] SER & Asterisk


> Thanks guys, thought SER had to 'register' to be able to use
> any Asterisk contexts.
> But just defining a new entry in the sip.conf with just context & ip
worked!
>
> But now i'm stumbling on another problem.. Asterisk seems to want
> to send the SIP udp packets directly to the SIP clients.
> In the case of a SIP user/client behind a NAT, this obviously doesn't
> work.
>
> SER is configured to use the wonderful RTPProxy + SER nathelper module,
> and this works flawlessly (using the rewritehostport function).
>
> But when I try to call a phone number on the PSTN network from a SIP
> client behind NAT, SER sends the invites to Asterisk, and Asterisk
> makes an outbound call to the phone number, the phone rings, but when
> the pstn user picks up the phone, no sound, and after a while (couple of
> seconds), the call is dropped.
> Asterisk spews out the following warning,
>   chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call
0c2f021a-7b4d-0133-d19b-7b1794308318 at 192.168.0.5 for seqno 29898 (Response)
> Tried searching on the voip-info wiki and mailinglists, but didn't find
> a way to force Asterisk to use a SIP proxy/SER.
>
> Any ideas ?
>
>
> On Fri, Jan 16, 2004 at 12:12:14AM -0800, Chris Albertson wrote:
> >
> > Yes, you can keep non-authorized SIP callers from accessing the
> > PSTN by setting up the .conf file "correctly" as below
> > but you can also
> > run a fire wall on the box that Asterisk runs on.  Firewall off
> > SIP ports except for if they come from your SER server.
> >
> >
> > --- Fran Boon <flavour at partyvibe.com> wrote:
> > > [ser]
> > > context=sip-legal
> > > host=y.y.y.y ; IP address of SER
> > >
> > > Se this Wiki page for more flesh of my (not yet fully working!)
> > > configs:
> > > http://voip-info.org/wiki-Asterisk+cisco+FXO
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users