Hi, all
I have Asterisk here and SIP phone sitting at another location.
Initially, I had problems registering the phone. Now I have added 'nat=yes'
for this phone in sip.conf and phone registers.
However, I can not make calls.
SIP debug shows that phone registers with public IP address of the
On 19:45, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote:
As one can see, public IP 147.10.78.157 is used at registration time, while
private IP 192.168.1.2 is used for communicating with phone.
[ext102]
type=user
nat=yes
host=dynamic
secret=ext102
context=default
[ext102]
type=peer
Thanks for suggestion.
Unfortunately did not work.
What does this option do anyway?
Rudolf
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 27, 2005 8:18 PM
Subject: Re: [Asterisk-Users] NAT/Routing problem
On 19:45
On 20:52, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote:
Thanks for suggestion.
Unfortunately did not work.
What does this option do anyway?
I cannot explain it as clear as the wiki.
have a look here:
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
--
Michiel van Baak
Hmm, is your asterisk server behind nat with port forwarded ports? If
so, have you tried adding this to your sip.conf?
[general]
externip=xxx.xxx.xxx.xxx; your external IP, provided by your ISP
localnet=192.168.0.0/255.255.255.0 ; your LAN
...
...
[ext102]
canreinvite=no
host=dynamic
Is your asterisk server behind nat with port forwarded ports? If
so, have you tried adding this to your sip.conf?
[general]
externip=xxx.xxx.xxx.xxx; your external IP, provided by your ISP
localnet=192.168.0.0/255.255.255.0 ; your LAN
...
...
[ext102]
canreinvite=no
host=dynamic
nat=yes
