[Asterisk-Users] Asterisk as a PSTN gateway for SER

[jerk face]

In sip.conf I have the following:

context=OUTGOING
autocreatepeer=yes

[Provider]
type=friend
username=XXXXX
secret=XXXXX
host=xxxxx.FakeProvider.com

So when Asterisk receives a call from SER it will
"autocreatepeer" and give access to the OUTGOING
context.



--- Jess Magnaye <jess at arretni.com> wrote:
> how did u setup your asterisk for this:
> 
> "I can also start a call through Asterisk to a VoIP
> provider, but there is a problem after the first
> ring:" 
> 
> 
> ----- Original Message ----- 
> From: "jerk face" <jerkface2098 at yahoo.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Monday, December 22, 2003 2:42 PM
> Subject: [Asterisk-Users] Asterisk as a PSTN gateway
> for SER
> 
> 
> > First off, here is what I want to do:
> > SIP Clients -> SER -> Asterisk -> VoIP provider
> > 
> > Where SER will handle communications between SIP
> > clients (since I would prefer that my SIP clients
> not
> > use all of my bandwidth)
> > Asterisk will handle calls to a VoIP provider
> > 
> > I have read that people have similar setups
> working,
> > but I have not seen any documentation of these
> setups.
> > 
> > So far, SIP Clients can talk to each other.
> > I can also start a call through Asterisk to a VoIP
> > provider, but there is a problem after the first
> ring:
> > 
> > Here is the output:
> > -- Executing Dial("SIP/-08114560",
> > "SIP/13239381067 at SIPprovider") in new stack
> > -- Called 13239381067 at SIPprovider
> > -- SIP/SIPprovider-5e0c is making progress passing
> it
> > to SIP/-08114560
> > -- SIP/SIPprovider-5e0c answered SIP/-08114560
> > -- Attempting native bridge of SIP/-08114560 and
> > SIP/SIPprovider-5e0c
> > 
> > I have tried this with my SIP client behind a NAT
> and
> > outside of a NAT, so I don't that is the problem.
> > I have also tried this with both IAX and SIP
> providers
> > and the problem is the same.  One ring, and then
> > silence.
> > 
> > Any thoughts?
> > 
> > Thank you for your time.
> > 
> > 
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