out* of india.
On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce wrote:
> There you go. This confirms that SIP signaling determines where the calls
> should go. I would take their word with a grain of salt specially with their
> whole support center our of India. No disrespect, but it is bad service
There you go. This confirms that SIP signaling determines where the calls
should go. I would take their word with a grain of salt specially with their
whole support center our of India. No disrespect, but it is bad service
overall.
-Bruce
On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp wrote:
> --
- "Tarek Sawah" wrote:
> we started with them two days ago .. and we are facing plenty of False
> Answer cases on several destinations although ppl said they have a
> policy against FAS..
> anyway i don't know i will be looking into another method to send the
> RTP to another server,
The IP
info
> Date: Sat, 10 Apr 2010 18:06:22 -0400
> From: bruceb...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Sending RTP media to a different server than
> SIP Signaling
>
> Oh, I see. I haven't done a lot of testing on
ital Systems
>
> CCNA, MCSE, RHCE, VoIP
>
> Syria: +963 944 618286
>
> USA: +1 347 562 2308
>
>
>
>
>
>
>
>
>
> > Date: Sat, 10 Apr 2010 15:50:52 -0400
> > From: bruceb...@gmail.com
> > To: asterisk-users
562 2308
> Date: Sat, 10 Apr 2010 15:50:52 -0400
> From: bruceb...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Sending RTP media to a different server than
> SIP Signaling
>
> Just a week ago, I
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
On Sat, Apr 10, 2010 at 8:50 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.
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