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

[Gary]

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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