On Tue, Dec 13, 2005 at 11:32:15PM -0500, Tom Rymes wrote:
On Dec 13, 2005, at 8:25 AM, Michael George wrote:
I have a similar problem with a client's system. They have * 1.0.x
behind a NAT with all the SIP phones on the local network. Their VoIP
provider is outside the NAT (a Metaswitch
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote:
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.
--- Jeffery
On Dec 13, 2005, at 8:25 AM, Michael George wrote:
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote:
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds
It can also be that the NAT is not truly SIP aware as it will create
some confusion if the IP address in the IP header is converted, while
the IP address in the SIP header is not. One cause would be that
messages are send to wrong address.
Jan
Wilson Pickett wrote:
i have an asterisk
i have an asterisk box behind the NAT ,when i try to
send calls through Sip to the voip provider server the
call is answered but in a one way calling,I hear the
voice of the other side just for 4 seconds and then
stop but the call do not hangup.
SOmetimes this can be due to the client using
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.
--- Jeffery Chen [EMAIL PROTECTED] wrote:
If your Astersik server behind NAT
can u paste your sip.conf general section,,?
there have another possible cause... the both side use different codecm and asterisk can not translaste it ...
-- Jeffery
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:i added these two lines to my general context ,butnothing happened the
Forward UDP Ports 1-2 to your asterisk box.
On 12/8/05, Jeffery Chen [EMAIL PROTECTED] wrote:
can u paste your sip.conf general section,,?
there have another possible cause... the both side use different codecm and asterisk can not translaste it ...
-- Jeffery
On 12/8/05, chawki hammoud
Hi list:
i have an asterisk box behind the NAT ,when i try to
send calls through Sip to the voip provider server the
call is answered but in a one way calling,I hear the
voice of the other side just for 4 seconds and then
stop but the call do not hangup.
my sip.conf is:
[voip provider]
type=peer
what type of NAT do you have? sync? full cone? cone restricted, port restricted?
any messages in asterisk verbose console?
best regardsOn 12/7/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi list:i have an asterisk box behind the NAT ,when i try tosend calls through Sip to the voip provider server
If your Astersik server behind NAT too, your need modify SIP.conf like this
externalIP= x.x.x.x
localnet= x.x.x.
hope this can help you
On 12/8/05, Moises Silva [EMAIL PROTECTED] wrote:
what type of NAT do you have? sync? full cone? cone restricted, port restricted?any messages in
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.
--- Jeffery Chen [EMAIL PROTECTED] wrote:
If your Astersik server behind NAT too, your need
modify SIP.conf like
this
Hi All,
Can anyone help me out here? I'm having some issues configuring my IPTables
firewall to properly NAT SIP and RTP packets to my asterisk server hiding
behind it.
Here are my current rules:
#Inbound SIP to HERMES
$IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 5060 -j DNAT --to
@lists.digium.com
Subject: [Asterisk-Users] SIP behind IPTables/NAT
Hi All,
Can anyone help me out here? I'm having some issues configuring my
IPTables firewall to properly NAT SIP and RTP packets to my asterisk
server hiding behind it.
Here are my current rules:
#Inbound SIP to HERMES
$IPTABLES
First off
Isn't RTP a TCP protocol? or am I over tierd again?
Secondly - unless several conditions are met (canreinvite=yes being
one of them) it (asterisk) will still proxy the connection. - Check
your dial statement for T's ie T and t - the wiki has a full list.
David
On 4/26/05, Ian
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