Hi Friends; Actually I would appreciate if Vivek can advise if the VPN resolved the RTP packets in the SIP Trunk between Asterisk and another softswitch? In other words, openvpn helpful in NAT cases in what exactly? As without VPN, I was able to establish a call but without voice or with complete noise (nothing understood) :) - So if NAT resolve this issue for the SIP Trunk, then I can proceed forward, as really now I do not have any other attempt to try.
>From the other side, I think that baji is talking about something else than the IP Trunk, he is talking about outbound (which is related to using an application to run an outside call, which is used usually in campaign in contact centers and so on), I think nthis case differs that placing a calls via IP Trunk or even outside call but the caller who will do it (and not the application). Lastly, Mr. Amit helped me when he gave me a configuration to be done for the SIP Trunk, as in his method, I did not register on the softswitch, I send directly, and the connectioned succeed, but as I said: with complete voice (actually nothing understood, i feel it is complete RTP situation), the test was by letting Asterisk behind NAT (private IP) and sending to a softswitch in anther country has a public IP address. Is it NAT issue, so VPN can resolve? Note: anyone knows if h323 works better in the IP trunk? Regards Bilal ---------------------------------- yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti <[EMAIL PROTECTED]> wrote: > > after a copious loss of follicles :-), I finally got outbound working. > > Basically the channel statement in the call file needs to have the > number to be called. For eg., in test.call format the statement > as follows : > > Channel: SIP/3012345678@<your-sip-provider> > > And there is no need for a DIAL statement in extensions.conf > unless you need to dial an additional number / extension. > > Then in sip.conf you need a para that matches <your-sip-provider> > with the relevant auth info. > > These two wiki pages, they were very helpful in figuring out a > solution to the problem : > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message > > hth, > > -baji. > > -- > > On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > > > I have the same problem. > > > > I trying with more 4 SIP providers, the account is registering, receive > > inboud calls, but can`t make outbound calls for "congestion". > > > > Can be the out call id the problem? > > > > Thanks > > Gabriel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
