[asterisk-users] Dropped Calls

[asterisk card support]

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: brent at texascountrytitle.com
> To: asterisk-users at lists.digium.com
> 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.
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100407/9b90d165/attachment.htm