[Asterisk-Users] Canreinvite

Il Neofita asteriskmail at gmail.com
Mon Jun 19 03:40:21 MST 2006


I will try your suggestion and I will let you know. Thank you

On 6/18/06, Philippe Lindheimer <p_lindheimer at yahoo.com> wrote:
>
> How have you confirmed that they did not reinvite? The channels are still
> controlled by Asterisk (sip signalling), it is the rtp streams that go
> direct. You can do a sip show channel 146b518a4cd on the specific channel to
> see where the rtp streams are going. Or ... if this is the only active
> channel on the box, just do a rtp debug. If the rtp stream is going through
> asterisk, it will be very obvious. If not, you won't see a constant flow of
> rtp debug messages going on.
>
> p
>
>
> From: "Il Neofita" <asteriskmail at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Date: Sun, 18 Jun 2006 05:01:20 -0400
> Subject: Re: [Asterisk-Users] Canreinvite
>
>
>
> This is the dial in extensions
> exten => _40001,1,Dial(SIP/40001,30)
> exten => _40002,1,Dial(SIP/40002,30)
>
> From: "Il Neofita" <asteriskmail at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Date: Sun, 18 Jun 2006 05:22:35 -0400
> Subject: Re: [Asterisk-Users] Canreinvite
>
>
> cosa vedo a console
>
>     -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack
>     -- Called 40002
>     -- SIP/40002-4753 is ringing
>     -- SIP/40002-4753 answered SIP/40001-3760
>     -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753
> srvlinux*CLI> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last
> Message
> 82.X2.XX3.X3     40002       146b518a4cd  00103/00000  alaw  No       Tx:
> ACK
> 82.X2.XX3.X3     40001       CBD1DB85-8B  00102/30987  alaw  No       Tx:
> ACK
> 2 active SIP channels
>
>
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