Well, unfortunately i did not dig much into "why/how it worked" with openvpn, but it did work for me with default setup.I think you may need to set constant ports instead of random ports.
Thanks, Vivek On 11/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: > > 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 >
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