hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default extension with SIP. is 600 and password 1234 so now i download xlite y configure it on the next way: user: 600 pass: 1234 auth user: 600 domain: pbxa.com nothing appers on the CLI, and after a 30 seconds i recieve a message on the xlite: Registration Error: 408- Request Timeout. (ping pbxa.com works fine), and btw, if i try with a user that doesnt exist (for example 601) on xlite i receive this on CLI: *CLI> [Apr 13 11:32:02] NOTICE[12896]: chan_sip.c:14530 handle_request_register: Registration from '"601"<sip:601@mrphones.net>' failed for '200.118.190.39' - No matching peer found i really dont get why i can register my SIP softphone, i try uninstalling and installing asterisk about 3 times and always is the same.... any ideas...? thanks you in advanced...
Hi again, i enabled SIP debug on the server. and appears this [Apr 13 11:49:57] <--- Transmitting (no NAT) to 192.168.0.100:19934 ---> 192.168.0.100 is my local ip address, there is the topology of the network: 72.232.33.66 -> asterisk server, 1gb/s direct internet access 232.32.76.11 -> my home internet adress 192.168.0.100 -> my laptop internal adress... (xlite cient) how i can solve this! please helpme =( 2007/4/13, Manolet Gmail <manolet@gmail.com>:> hi! > > First of all i want to tell i have a dedicated server on layeredtech > with direct internet connection and i currently dont use iptables, so > this is not about network configuration =). > > well so, i install asterisk-1.4.2 on my server, and next install > asterisk-gui from the digium repository. > > next i go to: > > http://pbxa.com:8088/asterisk/static/config/cfgbasic.html > > and install a default extension with SIP. is 600 and password 1234 > > so now i download xlite y configure it on the next way: > > user: 600 > pass: 1234 > auth user: 600 > domain: pbxa.com > > nothing appers on the CLI, and after a 30 seconds i recieve a message > on the xlite: Registration Error: 408- Request Timeout. > > (ping pbxa.com works fine), and btw, if i try with a user that doesnt > exist (for example 601) on xlite i receive this on CLI: > > *CLI> [Apr 13 11:32:02] NOTICE[12896]: chan_sip.c:14530 > handle_request_register: Registration from > '"601"<sip:601@mrphones.net>' failed for '200.118.190.39' - No > matching peer found > > > i really dont get why i can register my SIP softphone, i try > uninstalling and installing asterisk about 3 times and always is the > same.... any ideas...? > > thanks you in advanced... >
Hi Manolet, Can you provide your sip.conf? Thanks! -- Alex -- Alex Balashov <sasha@presidium.org>
of course, download it from here: http://contelecltda.com/sip.conf but i dont edit the sip.conf, is the default make samples sip.conf file. i just use the asterisk gui interface to add the user... 2007/4/13, Alex Balashov <abalashov@evaristesys.com>:> > Hi Manolet, > > Can you provide your sip.conf? > > Thanks! > > -- Alex > > -- > Alex Balashov <sasha@presidium.org> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:> of course, download it from here: > > http://contelecltda.com/sip.conf > > but i dont edit the sip.conf, is the default make samples sip.conf file. > i just use the asterisk gui interface to add the user...Well, then my conjecture would be that the GUI interface is broken, because there are no definitions for that or any other peer in there, nor hooks to include any other files generated by the GUI interface that might conceivably have them. Someone else have more insights? -- Alex Balashov <sasha@presidium.org>
mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? 2007/4/13, Alex Balashov <abalashov@evaristesys.com>:> On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: > > > of course, download it from here: > > > > http://contelecltda.com/sip.conf > > > > but i dont edit the sip.conf, is the default make samples sip.conf file. > > i just use the asterisk gui interface to add the user... > > Well, then my conjecture would be that the GUI interface is broken, > because there are no definitions for that or any other peer in there, > nor hooks to include any other files generated by the GUI interface > that might conceivably have them. > > Someone else have more insights? > > -- > Alex Balashov <sasha@presidium.org> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >