Re: [asterisk-users] externip nat audio sip trunk issue problem
On 02/01/2012 06:14 PM, Gabriel Ortiz Lour wrote: when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the packets to the correct SIP trunk IP but there is no response AT ALL from it... Use tcpdump -v (or wireshark) to look at the SIP packet contents as well as the IP headers. If possible, do this on the external interface of your firewall to see what's getting sent out to your peer. See if the values all appear to be sane. Compare the SIP session contents of packets with externip configured and packets without that setting. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] externip nat audio sip trunk issue problem
On Thu, Feb 2, 2012 at 8:51 AM, Gabriel Ortiz Lour wrote: > As soon as I activate the exterip/localnet config there is no response at > all, as if that IP address desappeared. > Any packets send to it simply get no response. > > I've considered being linux kernel routing issue, but since without the > exterip/localnet config it works OK I don't think this is the case. > > I could put the tcpdump/sip debug info here, but it would be like a > monologue, with only the packets being sent showing without any response. > But I can put it here if it helps... Change it to externip and localnat and enable sip debug in asterisk cli, then do a sip reload and post the cli output. > > Didn't anyone had this problem with this config option? > > > 2012/2/2 C F >> >> On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour >> wrote: >> > Hi all, >> > >> > I've tried search this problem on the list... no luck... >> > >> > The case is: >> > >> > without externip/localnet config on sip.conf [general] my SIP trunk >> > works, >> > but with no audio NAT problem (asterisk sends the private 192 address to >> > the >> > outside...) >> > >> > when I configure externip/localnet correctly my SIP trunk simply >> > disappear! >> > Checking the signalling with tcpdump shows me that Im sending the >> > packets to >> > the correct SIP trunk IP but there is no response AT ALL from it... >> >> Can you explain this? >> What do you mean no response? Is it registering? Do you have a debug >> output? >> >> > >> > Anyone had this problem? >> > >> > Thanks, >> > Gabriel >> > >> > -- >> > _ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > New to Asterisk? Join us for a live introductory webinar every Thurs: >> > http://www.asterisk.org/hello >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] externip nat audio sip trunk issue problem
As soon as I activate the exterip/localnet config there is no response at all, as if that IP address desappeared. Any packets send to it simply get no response. I've considered being linux kernel routing issue, but since without the exterip/localnet config it works OK I don't think this is the case. I could put the tcpdump/sip debug info here, but it would be like a monologue, with only the packets being sent showing without any response. But I can put it here if it helps... Didn't anyone had this problem with this config option? 2012/2/2 C F > On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour > wrote: > > Hi all, > > > > I've tried search this problem on the list... no luck... > > > > The case is: > > > > without externip/localnet config on sip.conf [general] my SIP trunk > works, > > but with no audio NAT problem (asterisk sends the private 192 address to > the > > outside...) > > > > when I configure externip/localnet correctly my SIP trunk simply > disappear! > > Checking the signalling with tcpdump shows me that Im sending the > packets to > > the correct SIP trunk IP but there is no response AT ALL from it... > > Can you explain this? > What do you mean no response? Is it registering? Do you have a debug > output? > > > > > Anyone had this problem? > > > > Thanks, > > Gabriel > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] externip nat audio sip trunk issue problem
On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour wrote: > Hi all, > > I've tried search this problem on the list... no luck... > > The case is: > > without externip/localnet config on sip.conf [general] my SIP trunk works, > but with no audio NAT problem (asterisk sends the private 192 address to the > outside...) > > when I configure externip/localnet correctly my SIP trunk simply disappear! > Checking the signalling with tcpdump shows me that Im sending the packets to > the correct SIP trunk IP but there is no response AT ALL from it... Can you explain this? What do you mean no response? Is it registering? Do you have a debug output? > > Anyone had this problem? > > Thanks, > Gabriel > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] externip nat audio sip trunk issue problem
Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the packets to the correct SIP trunk IP but there is no response AT ALL from it... Anyone had this problem? Thanks, Gabriel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users