Re: Asterisk progress

["UKHA" <ukha@xxxxxxx>]

Dan,

I think this may be ok for a single phone connection but you may run into
problems if someone tries to call another sip address at your site...

Andy

*********** REPLY SEPARATOR  ***********

On 07/05/2003 at 18:25 Dan wrote:

>For FWD you do not need a SIP enabled firewall (if NAT).
>For something else, is enough to open and forward port 5060 and two
ports
>for RTP (usualy 8000 and 8001).
>I have test it and it is enough.
>
>BR,
>Dan
>
>----- Original Message -----
>From: "UKHA" <ukha@xxxxxxx>
>To: <ukha_d@xxxxxxx>
>Sent: Wednesday, May 07, 2003 5:30 PM
>Subject: Re: [ukha_d] Asterisk progress
>
>
>> >
>> >Currently, I don't have any VoIP hardware and I am keen to get
some in
>to
>> >play, erm, evaluate. What hardware you would recommend or
suggest to
>start
>> >and from where? May be a couple of SIP phones and any
necessary PBX
>> >hardware.
>>
>> I forgot to add to my last message to you Li that you'll need
either some
>sort of SIP aware firewall or will have to open up lots of ports on
your
>existing one, the alternative is to stick the SIP proxy in your DMZ and
>give
>it an IP (non-nat)
>>
>> Andy
>>
>>
>>
>>
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>
>
>
>
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