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

[Jess Magnaye]

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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