[Asterisk-Users] TCP/UDP Ports used by Asterisk

[Dan Fernandez]

I believe it does handoff. I do it all the time and for me is extremely
helpful since I don´t want all the calls to go through my * box.

To do a handoff you just need to set canreinvite=yes on the sip.conf .

The problem I found then is with the ATA doing NAT since you need to  set
canreinvite=no, and therefore it won´t hand it off.




----- Original Message -----
From: "Gary" <gary at ausmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 24, 2003 1:16 PM
Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk


> I said wont/cant handoff
>
> which means it always goes thru asterisk, there asterisk can to any
> codec conversion.
>
> Hey, I might be wriong, this is only what I have seen, and I stand to
> be corrected.
>
> (I would luv SIP to handoff (lisk IAX)), would save me a heap of
> bandwidth !!
>
> On Sat, 24 May 2003 19:10:38 +0300, Dan wrote:
>
> >Hi Gary,
> >
> >> the SIP is using asterisk as a proxy, so therefore is wont/cant
> >> handoff.
> >Then how can it make codec conversion?
> >I have a Cisco 7960 hardware SIP phone (with G.711) and an X-Lite (with
> >GSM)... and they can talk each other through Asterisk..
> >
> >Dan
> >
> >----- Original Message -----
> >From: "Gary" <gary at ausmail.com>
> >To: <asterisk-users at lists.digium.com>
> >Sent: Saturday, May 24, 2003 7:05 PM
> >Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk
> >
> >
> >> from experience...
> >>
> >> the SIP is using asterisk as a proxy, so therefore is wont/cant
> >> handoff.
> >>
> >>
> >>
> >> On Sat, 24 May 2003 18:54:34 +0300, Dan wrote:
> >>
> >> >Hi all,
> >> >
> >> >I have my Asterisk behind a NAT router, but now it is configured to
put
> >that specific computer in DMZ (directly exposed to Internet).
> >> >I intend to disable this and to open just the used ports.
> >> >There is a list of TCP/UDP ports usd by Asterisk in order to connect
to
> >the outside world?
> >> >
> >> >One more question: When a call is established between an internal SIP
> >phone (in LAN) and a phone from another place outside my router/firewall,
> >using both the same codec (no conversion)... the call is still routed
> >through  the PBX or the PBX is used only for signaling and then a direct
> >connection between the two phones is established?
> >> >
> >> >I ask this because if the audio stream is passed through the PBX then
> >there is no need to open other ports on the firewall for the internal
> >phones.
> >> >
> >> >Thanks,
> >> >Dan
> >>
> >> .
> >>
> >>
> >>
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> >>
> >>
> >
> >
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> .
>
>
>
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