Depends if you're phone supports it, and you have reinvites etc enables
in *.
-d
At 03:17 PM 12/8/2003, you wrote:
Hi
all,
Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the
two sip phones( hard or soft) pas
> Brancaleoni Matteo wrote:
>
> > SIP control messages goes always through the server
> > (port 5060) , only RTP media streams is p2p .
> >
> > you can see RTP passing not p2p but by * server if:
> > * the phone doesn't supports reinvites
> > or
> > * set in sip.conf canreinvite=no in the user de
On Mon, Dec 08, 2003 at 10:20:43PM +0100, Brancaleoni Matteo wrote:
> SIP control messages goes always through the server
> (port 5060) , only RTP media streams is p2p .
>
> you can see RTP passing not p2p but by * server if:
> * the phone doesn't supports reinvites
> or
> * set in sip.conf canrei
Hello!
I think that's true. In older asterisk versions I saw such a "hand-over"
between 2 sip phones and asterisk. But with the current versions I can't
get it to work. I think you have to set "canreinvite=yes" at both
clients that this can work. Additionally both ends need to have a common
co
Brancaleoni Matteo wrote:
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf canreinvite=no in the user definition
or if the both ends have
You're right for pure SIP configurations , but Asterisk acts here as "media gateway"
and treats all of the media comm. e.g. to eventually communicate with other types
(like PSTN, H323, AIX, ... the voicemail app.)
Michael Devenijn
IT Manager
DKMA
Schaarbeeklei 636
B-1800 Vilvoorde
Tel.: +32
Wim Venneman wrote:
Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the
two sip phones( hard or soft) passing through the asterisk server (on
UDP layer)
Yes.
Isn't SIP a protocol that (after that it has est
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf canreinvite=no in the user definition
Matteo.
Il lun, 2003-12-08 alle 22:17, Wim Venne
