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@SIPprovider") in new stack -- Called 13239381067@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. __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. photos.yahoo.com
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@yahoo.com> To: <asterisk-users@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@SIPprovider") in new stack > -- Called 13239381067@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. > > > __________________________________ > Do you Yahoo!? > New Yahoo! Photos - easier uploading and sharing. > photos.yahoo.com > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users