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...really sure if this is the better way to do it and I can''t do as many "shorewall restarts" as Id like because of the users of the system, so I prefer to be as sure as I could before try anything. Many thanks for all -- Un Saludo. Javier Ramirez Molina Departamento Ingeniera Scancom jramirez@scancom.es Telefono: 952486557

...loc net:192.168.1.11 tcp 4662 DNAT loc net:192.168.1.11 udp 4665 DNAT loc net:192.168.1.11 udp 4672 What do you think? Any suggestion? -- Un Saludo. Javier Ramirez Molina Departamento Ingeniera Scancom jramirez@scancom.es Telefono: 952486557

Hello, everyone. I'm having a small problem when using asterisk with GUI. For every provider I create I have to set "insecure=invite,port" in users.conf. Is there a way to make it a default setting? Thanks in advance.

Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, when making a call, my asterisk box doesn't detect a ringback and I just hear silence until the other party picks up the phone. I've checked the SIP messages and they are ok (I'm getting 183 "session in progress"), so I guess I should be debugging the RTP packets. From then on

Hello, everyone. I'm reading about the asterisk new features. One is T.38 protocol support. I used faxes before with asterisk 1.2 and everything was working quite well. Could anyone explain what have changed in the way faxes are handled. Another thing is, in order for asterisk to work over T.38 with my fax machine do I also need a T.38 support from my ATA and my SIP provider? Thanks in

Hello, everyone. I'd like to ask how does asterisk 1.4 with GUI register itself at the provider's end (when I mark a checkbox 'register' while creating a "Service Provider"). Before I used to write something like: register => 924980111:xxxxxxx@sip.provider.com/924980111 in sip.conf. Having that line, asterisk would execute exten => 924980111,1,..... when receiving

Hello, everyone. I've installed asterisk SVN-branch-1.4-r62942 and every time I reload asterisk I get this in CLI: -- Reloading module 'app_playback.so' (Sound File Playback Application) [May 3 20:04:26] NOTICE[13892]: app_playback.c:455 reload: Reloading say.conf == Parsing '/etc/asterisk/say.conf': Found [May 3 20:04:26] WARNING[13879]: frame.c:1289

Hello, everyone I'm looking for a way to find out if there is a device registered on a particular extension from dialplan. Does anyone know how to do that? Thanks in advance.

Hi, everyone I'm investigating if I could give asterisk CLI access to one of our clients. If I add that user to asterisk group and set his shell to /usr/sbin/rasterisk, is there a possibility for a user to brake our of asterisk CLI to normal shell? Thanks in advance

Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: piiiiiiiiiiii---piiiiiiiiiiiiii....) and in almost all cases that is true. However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the

Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, "default" is always being used. The output of "sip show peers" shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the "default" context.