an B. Christensen
> Senior technician
> Ibidium AS
> http://www.ibidium.no/
>
> - Original Message -
> *From:* Bruce B
> *To:* Asterisk Users Mailing List - Non-Commercial
> Discussion
> *Sent:* Tuesday, January 11, 2011 4:37 PM
> *Subject:* Re: [asterisk-users]
tp://doc.pfsense.org/index.php/VoIP_Configuration
http://en.wikipedia.org/wiki/Application-level_gateway
With kind regards,
Pan
From: Bruce B
Sent: Tuesday, January 11, 2011 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Do I need
Hi,
> At least
> that is my understanding of NAT. The provider should see me trying to
> register from the same IP with multiple different ports (high number
> ports; not talking about 5060 as this is outbound and not inbound) and
> should be able to differentiate between SIP packets coming from v
/VoIP_Configuration
> http://en.wikipedia.org/wiki/Application-level_gateway
>
> With kind regards,
> Pan
>
> *From:* Bruce B
> *Sent:* Tuesday, January 11, 2011 8:58 AM
> *To:* Asterisk Users Mailing List - Non-Commercial
> Discussion
> *Subject:* [asterisk-users
- Non-Commercial Discussion
Subject: [asterisk-users] Do I need a sip proxy?
Hi Everyone,
I am running multiple instances of Asterisk in Proxmox and so far I had one
central Asterisk feeding all others with trunks from one provider. Now, I want
to connect each Asterisk server directly to the
Hi Everyone,
I am running multiple instances of Asterisk in Proxmox and so far I had one
central Asterisk feeding all others with trunks from one provider. Now, I
want to connect each Asterisk server directly to the provider. Based on my
understanding, each connection made to the provider port 506
No, you don't necessarily need a SIP proxy for this. Furthermore, while
a SIP proxy might assist with certain SIP-level reachability issues, it
will do nothing for the actual audio (media) if there are NAT issues
that prevent that from getting through.
As the other reply said, "this isn't work
On Wed, May 20, 2009 at 1:50 PM, Tim Nelson wrote:
> Could you elaborate a bit more?
> What isn't 'working out to well'?
> Are you getting failed calls? One way or no audio?
Sorry for the lack of information. I posted in a bit of haste.
Initially it was failed calls, or not being able to registe
- "Jonathan Moore" wrote:
> I've got an Asterisk server, and several SIP phones behind our router
> here. Things are working just perfectly inside the network, just as
> the should.
>
> However, I'm not trying to configure my asterisk server to talk with
> SIP services outside our network, o
I've got an Asterisk server, and several SIP phones behind our router
here. Things are working just perfectly inside the network, just as
the should.
However, I'm not trying to configure my asterisk server to talk with
SIP services outside our network, once such example is my gizmo
project accoun
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