[asterisk-users] Sending RTP media to a different server than SIP Signaling

[bruce bruce]

Oh, I see. I haven't done a lot of testing on this new IP since the change
of gateways happened but I did Canada calls and they go fine. However, this
exact provider lies down to their teeth when it comes to problems of call
quality and calls not routing. They never accept faults. They even have
problems sending calls to Canada and USA. They failed to pass calls to India
as well over times. I had a funny issue where they were blocking one
specific area code in USA without even telling us. It was just a regular
area code. They told me it was blocked but I know it was a lie because they
wanted to cover their a$$ as the route was down and it wasn't blocked.

I doubt the problem is with sending calls to different media gateway as I
think SIP signals take care of that. Just like canreinvite feature. But I
reserve the right to be wrong.

-Bruce

On Sat, Apr 10, 2010 at 4:45 PM, Tarek Sawah <tareksawah at hotmail.com> wrote:

>
>
> you got the name EXACTLY!
> i already am doing what you suggest but facing problems with some
> destinations and they claim that the problem is with my Asterisk server not
> their routes!
>
>
>
> --
> AHD Tarek Sawah
>
> Integrated Digital Systems
>
> CCNA, MCSE, RHCE, VoIP
>
> Syria: +963 944 618286
>
> USA: +1 347 562 2308
>
>
>
>
>
>
>
>
> ________________________________
> > Date: Sat, 10 Apr 2010 15:50:52 -0400
> > From: bruceb444 at gmail.com
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [asterisk-users] Sending RTP media to a different server
> than SIP Signaling
> >
> > Just a week ago, I have been in the same situation. Provider was changing
> from Cisco gateways to I think Nextone and hence provided me many IPs.
> >
> > I found out that the media IPs don't matter and just played around with
> my NAT settings and all calls can go through just fine by using simply:
> >
> >
> > host=111.111.111.111
> >
> > and the 111.111.111.111 is just their SIP signaling IP. Their gateway
> will then transfer asterisk to proper gateways for media.
> >
> > Just give it a try; it should work. But my efforts on finding anything
> regarding this failed on Google as well.
> >
> >
> > P.S. the voip provider name starts with a T and end with A.
> >
> > Regards,
> > Bruce
> >
> > On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah> wrote:
> >
> >
> >
> > Greetings list
> >
> > i'm trying to connect with a VoIP provider for termination.. and they
> have offered us three servers to connect with
> >
> > one SIP Signaling server and Two Media servers ..
> >
> > googled for a week and didn't find a way to do this.. so my question. is
> it possible to be done?
> >
> > Asterisk server 1.4.26.3
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
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