hi:
how about the codecs?
Best wishes!
Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7,
elastix, trixbox support.
website:www.cnasterisk.com, www.voip88.com
> Date: Wed, 31 Mar 2010 17:20:30 -0500
> From: [email protected]
> To: [email protected]
> Subject: Re: [asterisk-users] Dropped Calls
>
> On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:
> >
> > Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
> >
> I was suspecting something with either rtptimeout or sip registration
> timeout, but I'm not sure what.
>
> --
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