Re: [asterisk-users] SIP REGISTRATION TIME OUT
Hi, now i can log in ok on my xlite, somebody calls me and everythink its okey. i hear and the caller hear. (the pc with the xlite have DMZ). But now i close xlite and put the same extension on a grandstream 286 (dont have DMZ). When somebody calls me the caller can hear me. but i cant hear! whats the problem? with other providers i can talk using my grandstream 286 without give it dmz or changing the configuration on my router. i hopes somebody can help me! 2007/4/14, dave cantera [EMAIL PROTECTED]: hello, I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I loaded 1.4 over *NOW because the gui regenerates files that, well, don't seem to work very well. it seems to me the gui creates the users.conf file, and then a script creates or uses the users.conf to create the dialplan... here is the users.conf file from *NOW... as you can see, this file does not conform to either sip.conf or extensions.conf, so that is my reasoning that it is source for some other generator... daveC ;! ;! Automatically generated configuration file ;! Filename: users.conf (/etc/asterisk/users.conf) ;! Generator: Manager ;! Creation Date: Sun Jan 21 15:41:42 2007 ;! [general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 host = dynamic localextenlength = 4 ;[6000] ;fullname = Joe User ;email = [EMAIL PROTECTED] ;secret = 1234 ;zapchan = 1 ;hasvoicemail = yes ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international Nicholas Campion wrote: The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 http://192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.conf in the general section. Give that a try and see what your result is. Nick On 4/13/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: 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?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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 ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
hi, to get it work i change under sip.conf nat: route Allow RTP reinvite:update with that i can hear, without dmz... but... why? 2007/4/19, Manolet Gmail [EMAIL PROTECTED]: Hi, now i can log in ok on my xlite, somebody calls me and everythink its okey. i hear and the caller hear. (the pc with the xlite have DMZ). But now i close xlite and put the same extension on a grandstream 286 (dont have DMZ). When somebody calls me the caller can hear me. but i cant hear! whats the problem? with other providers i can talk using my grandstream 286 without give it dmz or changing the configuration on my router. i hopes somebody can help me! 2007/4/14, dave cantera [EMAIL PROTECTED]: hello, I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I loaded 1.4 over *NOW because the gui regenerates files that, well, don't seem to work very well. it seems to me the gui creates the users.conf file, and then a script creates or uses the users.conf to create the dialplan... here is the users.conf file from *NOW... as you can see, this file does not conform to either sip.conf or extensions.conf, so that is my reasoning that it is source for some other generator... daveC ;! ;! Automatically generated configuration file ;! Filename: users.conf (/etc/asterisk/users.conf) ;! Generator: Manager ;! Creation Date: Sun Jan 21 15:41:42 2007 ;! [general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 host = dynamic localextenlength = 4 ;[6000] ;fullname = Joe User ;email = [EMAIL PROTECTED] ;secret = 1234 ;zapchan = 1 ;hasvoicemail = yes ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international Nicholas Campion wrote: The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 http://192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.conf in the general section. Give that a try and see what your result is. Nick On 4/13/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: 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?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
hello, I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I loaded 1.4 over *NOW because the gui regenerates files that, well, don't seem to work very well. it seems to me the gui creates the users.conf file, and then a script creates or uses the users.conf to create the dialplan... here is the users.conf file from *NOW... as you can see, this file does not conform to either sip.conf or extensions.conf, so that is my reasoning that it is source for some other generator... daveC ;! ;! Automatically generated configuration file ;! Filename: users.conf (/etc/asterisk/users.conf) ;! Generator: Manager ;! Creation Date: Sun Jan 21 15:41:42 2007 ;! [general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 host = dynamic localextenlength = 4 ;[6000] ;fullname = Joe User ;email = [EMAIL PROTECTED] ;secret = 1234 ;zapchan = 1 ;hasvoicemail = yes ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international Nicholas Campion wrote: The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 http://192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.conf in the general section. Give that a try and see what your result is. Nick On 4/13/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: 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?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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
[asterisk-users] SIP REGISTRATION TIME OUT
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 '601sip:[EMAIL PROTECTED]' 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... ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
Hi Manolet, Can you provide your sip.conf? Thanks! -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
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 [EMAIL PROTECTED]: Hi Manolet, Can you provide your sip.conf? Thanks! -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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 ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
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 [EMAIL PROTECTED] ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
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 [EMAIL PROTECTED]: 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 [EMAIL PROTECTED] ___ --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 ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: 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?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] ___ --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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.confin the general section. Give that a try and see what your result is. Nick On 4/13/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: 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?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] ___ --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 ___ --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
