On 01/11/2012 12:09 PM, Bryant Zimmerman wrote:
*From*: "Steve Davies"
*Sent*: Wednesday, January 11, 2012 12:51 PM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
*Subject*: Re: [ast
From: "Steve Davies"
Sent: Wednesday, January 11, 2012 12:51 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] SIP and NAT best practices since recent
changes?
On 11 January
On 11 January 2012 15:43, Kevin P. Fleming wrote:
> On 01/11/2012 05:29 AM, Steve Davies wrote:
>>
>> Hi,
>>
>> Since the recent update to the NAT configuration options and defaults
>> in chan_sip.so, I am interested in any SIP/NAT best practices advice.
>>
>> What I've always done in the past is:
On 01/11/2012 05:29 AM, Steve Davies wrote:
Hi,
Since the recent update to the NAT configuration options and defaults
in chan_sip.so, I am interested in any SIP/NAT best practices advice.
What I've always done in the past is:
Global: nat=no
SIP handsets that are local: nat=no
SIP handsets that
Hi,
Since the recent update to the NAT configuration options and defaults
in chan_sip.so, I am interested in any SIP/NAT best practices advice.
What I've always done in the past is:
Global: nat=no
SIP handsets that are local: nat=no
SIP handsets that are remote: nat=yes
ITSP SIP trunks: nat=yes
Apart of what everyone writes with the NAT=YES I would suggest using
canreinvite=no as well as normally asterisk cans the reinvite and this
might cause the audio not to get through the NAT and cause dead air for
the users specially if the users are behind 2 seperate NAT servers eg.
different p
Leo Ann Boon <> wrote on Sunday, 22 January 2006 4:32 PM:
> Trevor G. Hammonds wrote:
>
>> While I have not used siproxd, I have read a bit about it. From my
>> understanding of the docs, the local SIP agents register to siproxd,
>> but siproxd does not register to Asterisk. So the calls will
>
Trevor G. Hammonds wrote:
While I have not used siproxd, I have read a bit about it. From my
understanding of the docs, the local SIP agents register to siproxd, but
siproxd does not register to Asterisk. So the calls will traverse the NAT
properly, but features like MWI will not work in this
Leo Ann Boon wrote on Saturday, 21 January 2006 6:21 PM:
> Trevor G. Hammonds wrote:
>
>> How about when you have four or five SIP devices at a single
>> location? Do you manually assign each phone a separate port and add
>> firewall/router rules? I am looking for an inexpensive device or
>> met
I thing, that configuring nat device/firewall at consumer site isn't
always possible, thus simplest (but not optimal) way is to configure
phone in sip.conf as nat=yes & canreinvite=no, this should work in most
cases even if multiple phones are behind same nat, like adsl router.
disadvatage is, t
Trevor G. Hammonds wrote:
How about when you have four or five SIP devices at a single location? Do
you manually assign each phone a separate port and add firewall/router
rules? I am looking for an inexpensive device or method that will allow
this happen automatically. Rather than going that
How about when you have four or five SIP devices at a single location? Do
you manually assign each phone a separate port and add firewall/router
rules? I am looking for an inexpensive device or method that will allow
this happen automatically. Rather than going that route, my current
solution is
Most often the simple addition of nat=yes in the relevant sip.conf
stanza is all that's required to make a remote SIP phone work from
behind a firewall.
for example
[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
nat=yes
I've been running 4 remote SIP phones across the internet
Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk
server somewhere where there was no NAT for the * box that the SIP phones
wouldn't create any issues.
How do you people with Hosted PBX handle the deployment of SIP phones behind
NAT firewalls? Is it just elbow grease an
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