Re: [Asterisk-Users] Sip behind the NAT

2005-12-15 Thread Michael George
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-13 Thread Michael George
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-13 Thread Tom Rymes
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-12 Thread [EMAIL PROTECTED]
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-09 Thread Wilson Pickett
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-09 Thread Tom Rymes
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-08 Thread Jeffery Chen
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-08 Thread Bharath
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

[Asterisk-Users] Sip behind the NAT

2005-12-07 Thread 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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-07 Thread Moises Silva
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-07 Thread Jeffery Chen
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

Re: [Asterisk-Users] Sip behind the NAT

2005-12-07 Thread chawki hammoud
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

[Asterisk-Users] SIP behind IPTables/NAT

2005-04-26 Thread Ian Pattison
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

RE: [Asterisk-Users] SIP behind IPTables/NAT

2005-04-26 Thread Johan Akerstrom
@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

Re: [Asterisk-Users] SIP behind IPTables/NAT

2005-04-26 Thread David John Walsh
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