Wiley Siler
2005-Jun-02 09:51 UTC
[Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working
Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing this project or not. If someone has it working please just let me know. -- Thanks, Lance Grover _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Wiley Siler
2005-Jun-02 09:53 UTC
[Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working
<sig> type too fast and shift + enter will send the email early... OK. Anyway... The parameters below are important for the issue you have. The wiki covers this under the sip section www.voip-info.org W -----Original Message----- From: Wiley Siler Sent: Thursday, June 02, 2005 9:51 AM To: 'Lance Grover'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing this project or not. If someone has it working please just let me know. -- Thanks, Lance Grover _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Lance Grover
2005-Jun-02 10:56 UTC
[Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working
On 6/2/05, Wiley Siler <wsiler@education2020.com> wrote:> Have you configured your sip.conf to use these aprameters under General? > > ;externip=66.213.227.66 > ;localnet=192.168.1.0 > ;localmask=255.255.255.0 >Yes, I have those configured along with nat and qualify under each phone config. -- Thanks, Lance Grover
Lance Grover
2005-Jun-02 11:02 UTC
[Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working
On 6/2/05, Wiley Siler <wsiler@education2020.com> wrote:> <sig> type too fast and shift + enter will send the email early... > > OK. Anyway... The parameters below are important for the issue you > have. > The wiki covers this under the sip section www.voip-info.org > > WI have spent a lot of time on www.voip-info.org, and several other sites, but thanks all the same. So have you gotten this setup to work for you then? -- Thanks, Lance Grover