[asterisk-users] Everyone is busy/congested: IP Trunk

[Vivek Shrivastava]

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 <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
> trunk?
>
> Regards
> Bilal
>
> ----------------------------------
> yeah i found openvpn helpful in NAT cases.
>
> -Vivek
>
>
> On 11/6/07, Baji Panchumarti
> <baji.panchumarti at gmail.com> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071110/ec2b51bc/attachment.htm