Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9)
Hi
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Joshua Colp
Inviato: venerd? 28 luglio 2006 12.54
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] Canreinvite
----- Original Message -----
From: Giordano Grandis
[mailto:g.grandis@invidea.it]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com]
Sent:
Fri, 28 Jul 2006 07:01:08 -0300
Subject: [asterisk-users] Canreinvite
> How can I check if SIP re-invite is really working ?
If you do a sip debug you should see two INVITEs to each side after the call is
established with the IP address of the GXP2000 in the SDP. You can also run rtp
debug to see if the RTP audio stream is running through Asterisk.
> I'm trying it with two grandstream gxp2000.
>
> Thanks
>
Joshua Colp
Digium
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