[asterisk-users] asterisk is not sip proxy

[Jaswinder Singh]

Asterisk is not a sip proxy but it *can* partly act as a sip proxy if
reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk
connects 2 end points directly and does signalling between them .
Asterisk is a PBX now suppose u need to record all calls ..do conferencing
stuff  then rtp stream need to pass from asterisk (  openser cant do this
bcoz it just connects 2 endpoints  and only does signalling ) .. If you do
canreinvite=yes in sip.conf for both peers then asterisk does only
signalling ( also dial command should not have transfer parameters tT .. ) .
If both peers are behind NAT then asterisk reinvites may not work properly .

On 23/07/07, bilal ghayyad <bilmar_gh at yahoo.com> wrote:
>
> Dear Edgar;
>
> I am little bit confused, do u mean that asterisk does
> not work in that way:
>
> RTP (media) to be from the sournce to the destination
> directly while signaling to be via asterisk?
>
> So, what he parameter canreinvite is doing?
>
> Regards,
> ------------
> ITS
> Ip Telephony and Contact Center Engineer
> Eng. Bilal Ghayad
> Mobile: 00965 9849460
>
>
>
>
> Hello Asteriskers,
>
> I'm confused about why Asterisk is not a SIP proxy and
> why exactly
> this can affect the performance of a large Asterisk
> system.
>
> I know that Asterisk acts as a useragent endpoint, but
> my doubt is why
> exactly Asterisk could overload the call flow if the
> RTP voice stream
> goes from the caller to the called party.
>
> Does someone know how many calls or pencentaje that
> could handle a SER
> or OpenSER in comparison with Asterisk?
>
>
>
>
>
>
>
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