Hello averybody,
In a no natted environment if I letnat=yes on sip.conf it would cause some
thing bad or it is irrelevant ? Anybody know ?
thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda --
On 07/26/2011 09:19 AM, Flavio Miranda wrote:
In a no natted environment if I letnat=yes on sip.conf it would
cause some thing bad or it is irrelevant ? Anybody know ?
There is no harm unless the endpoint you are dealing with does not do
symmetric RTP. The nat=yes option assumes that it is
@lists.digium.com
Subject: Re: [asterisk-users] NAT yes
On 07/26/2011 09:19 AM, Flavio Miranda wrote:
In a no natted environment if I letnat=yes on sip.conf it would
cause some thing bad or it is irrelevant ? Anybody know ?
There is no harm unless the endpoint you are dealing with does not do
On 07/26/2011 09:29 AM, Flavio Miranda wrote:
I am experiencing some one-way audio, that's the reason of the
questions!
There are many possible reasons for it, but asymmetric RTP + 'nat=yes'
may be one of them.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite
Also consider the setting localnet in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Tuesday, July 26, 2011 9:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users
On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Dear * users,
in your opinion, when using a * as a public server, is good practice
enabling nat=yes in sip.conf for all the peers?
Can anyone imagine a scenario when enabling this parameter (even for
stot...@asteriskhelpdesk.com
Subject: Re: [asterisk-users] Nat=yes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, April 24, 2011, 2:13 PM
On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Dear
On Sun, Apr 24, 2011 at 5:55 AM, Muhammad Ali ali_...@yahoo.com wrote:
Hi,
When NAT = YES, Asterisk server will extract IP from the network layer.
When Nat = No, the Asterisk server will respond to the IP in the SIP
header. Am I right?
May be such type of options can be helpful for SIP
for SIP application developers rather then end customers.
Have a good time.
Regards
--- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote:
From: Steve Totaro stot...@asteriskhelpdesk.com
Subject: Re: [asterisk-users] Nat=yes
To: Asterisk Users Mailing List - Non-Commercial Discussion
check this
http://www.voip-info.org/wiki/view/Asterisk+sip+nat
On Thu, Apr 21, 2011 at 2:12 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Dear * users,
in your opinion, when using a * as a public server, is good practice
enabling nat=yes in sip.conf for all the peers?
Can
Dear Benjamin;
So in that case, when we set nat = yes? For what we do
this?
C F, I have nat=yes set by default for all my
extensions(with
canreinvite=no). And things work fine.
Bilal, about Asterisk sending packets to
public/private :
Asterisk will send packets to the public IP advertised
by
So I'll rephrase to some devices will not operate properly, since
after your message I am assuming that you tested this with most
devices.
On 9/10/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
C F, I have nat=yes set by default for all my extensions(with
canreinvite=no). And things work fine.
C F wrote:
BTW, AFAIK, there is no such thing as host=static it's either dynamic
or an IP/Name.
Yeah, I learned that the hard way. I had only set up dynamic devices
for a couple of months, and the first time I had reason to set up a
device with a static IP, I just assumed that
Hi List;
If I set nat=yes, then asterisk will send the packets
to the public IP address or to the private IP address
(which will be for the endpoint that is behind the
nating)?
And by setting the nat=yes, then what exactly will be
ignored at asterisk side when reading the
registeration messages
If you set yes then asterisk assumes that the address its coming from
is not the same as the UA thinks it is. most devices will not operate
properly if set to yes when they are in fact local.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
If I set nat=yes, then asterisk will send
BTW, AFAIK, there is no such thing as host=static it's either dynamic
or an IP/Name.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
If I set nat=yes, then asterisk will send the packets
to the public IP address or to the private IP address
(which will be for the endpoint that is
C F, I have nat=yes set by default for all my extensions(with
canreinvite=no). And things work fine.
Bilal, about Asterisk sending packets to public/private :
Asterisk will send packets to the public IP advertised by the msg/recv
from address. It is the NAT's headache on the endpoints network
Looking for some feedback on whether nat=yes and qualify=yes
will provide a workable solution in many cases?
The * server is on a public address, no NAT, the UAs (sipura,
linksys, polycom) are behind various types of NAT.
Obviously port mapping in the NAT device works, but what
about
Sipura works, I never tried linksys, Polycom might and might not work.
On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote:
Looking for some feedback on whether nat=yes and qualify=yes will provide a
workable solution in many cases?
The * server is on a public address, no NAT, the UAs
-Users] nat=yes and qualify=yes viable NAT
solutions?
Sipura works, I never tried linksys, Polycom might and might not work.
On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote:
Looking for some feedback on whether nat=yes and qualify=yes will
provide a
workable solution in many cases
Of C F
Sent: Sunday, February 26, 2006 7:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] nat=yes and qualify=yes viable NAT
solutions?
Sipura works, I never tried linksys, Polycom might and might not work.
On 2/26/06, Damon Estep [EMAIL PROTECTED
Don't have a clue why changing your settings causes the phone call to
fail, but obviously it needs to be investigated. The keyword is not
supported in code, therefore something else is impacting your config.
OK, so I have a nonexistant line in my settings. Why then when
canreinvite= is a valid option. reinvite= is not a valid option.
Rich Adamson wrote:
Don't have a clue why changing your settings causes the phone call to
fail, but obviously it needs to be investigated. The keyword is not
supported in code, therefore something else is impacting your config.
Title: NAT=YES
Good morning
Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP.
Configuration Info:
I have all users in
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf
Mark
Klint, Peter wrote:
Good morning
Does anyone have experience with NAT=YES? I have the following
configuration and am a bit confused as to why the Asterisk server
initially sends out RTP to the remote host private IP
FYI, there is no such thing as reinvite. Someone started using
that in postings a long time ago and a few people keep posting
it, but it doesn't exist.
(Check /usr/src/astersik/configs/sip.conf.sample)
Add canreinvite=no and reinvite=no to the relevant stanza in
Mark Phillips wrote:
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf
Anyone that tells you to use reinvite= is confused. The option does not
exist (check the source code if you don't believe me). reinvite= is one
of the many Asterisk Urban Myths.
--
Eric Wieling *
OK, so I have a nonexistant line in my settings. Why then when I remove
it does my phone call fail?
Rich Adamson wrote:
FYI, there is no such thing as reinvite. Someone started using
that in postings a long time ago and a few people keep posting
it, but it doesn't exist.
(Check
I am trying to use asterisk as a gateway between SER and the PSTN.
Should the nat=yes config work with these sip.conf settings ?asterisk is
trying to send it's response
back to the private IP.
[general]
context=OUTGOING
autocreatepeer=yes
[Provider]
type=friend
username=X
Hmm. this rings a bell, try putting nat=yes in your sip.conf, I think that fixed
the problem for me. (Or was the the login timed out thing? *shrug*)
The manual is not very clear on what happens with nat=yes in sip.conf.
Anyone here that could write a simple explanation of this option?
/O
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