[Asterisk-Users] How to tell if RTP stream is has been reinvited?

[C F]

rtp debug

On 5/15/06, Philippe Lindheimer <p_lindheimer at yahoo.com> wrote:
>
> I do a sip debug on the appropriate channel or IP address and look at the
> SIP messages. Would be great if there were an easier way though?
>
> p
>
>
> From: "Brent Torrenga" <lists at torrenga.com>
> To: <asterisk-users at lists.digium.com>
> Date: Mon, 15 May 2006 12:52:19 -0500
> Subject: [Asterisk-Users] How to tell if RTP stream is has been reinvited?
>
> Howdy,
>
> How can you tell if RTP traffic has been reinvited/is bypassing an * server?
>
>
> Sincerely,
>
> Brent A. Torrenga
> brent.torrenga at torrenga.com
>
> Torrenga Engineering, Inc.
> 907 Ridge Road
> Munster, Indiana 46321-1771
>
> +1 219 836 8918 x325 Voice
> +1 219 836 1138 Facsimile
> www.torrenga.com
>
>
>
>
>  ________________________________
> New Yahoo! Messenger with Voice. Call regular phones from your PC and save
> big.
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>