[asterisk-users] Everyone is busy/congested: IP Trunk
vivshrivastava at gmail.com
Sat Nov 10 05:41:18 CST 2007
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.
On 11/9/07, bilal ghayyad <bilmar_gh at yahoo.com> 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
> yeah i found openvpn helpful in NAT cases.
> On 11/6/07, Baji Panchumarti
> <baji.panchumarti at gmail.com> wrote:
> > after a copious loss of follicles :-), I finally got
> > 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
> > unless you need to dial an additional number /
> > Then in sip.conf you need a para that matches
> > with the relevant auth info.
> > These two wiki pages, they were very helpful in
> figuring out a
> > solution to the problem :
> > 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
> > > inboud calls, but can`t make outbound calls for
> > >
> > > Can be the out call id the problem?
> > >
> > > Thanks
> > > Gabriel
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