R: [asterisk-users] Canreinvite

Fri Jul 28 08:02:01 MST 2006

Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9)

Hi

-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Joshua Colp
Inviato: venerdì 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 at invidea.it]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:asterisk-users at 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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