search for: pbrufal

...e with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you know what is happening and how can I fix it? It's an only SIP system, no fxo/fxs cards. Thanks in advance. -- Paco Brufal <pbrufal@servitux.es> ServiTux Servicios Inform?ticos S.L. Tel. 966 160 600 / Fax. 966 160 601

...tended transfer option (x and X). I see a patch here: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 but this patch doesn't applies well in cvs version or 1.0.5. How can I make attended transfers work? It's a feature really needed :( Thanks in advance :) -- Paco Brufal <pbrufal@servitux.com> Servitux Servicios Inform?ticos S.L. http://www.servitux.com/ Telefono: 966090600 Fax: 966090601

...is no conversation, and 3 minutes later, it hangs up the line. The solution is to restart Asterisk. I can't access to the configuration of Nokia32 because I don't know the password. Someone can tell me what is happening and how to solve this? Thanks in advance. -- Paco Brufal <pbrufal@servitux.es> ServiTux Servicios Inform?ticos S.L. Tel. 966 160 600 / Fax. 966 160 601

...8.1 and one quadbri. I have installed the latest bristuff patches (0.3.0-PRE-1s). The system works fine, but when I do a "reboot", the system hangs unloading module "qozap". Is there any known problem with latest 2.6 kernels and qozap module? Thanks. -- Paco Brufal <pbrufal@servitux.es> ServiTux Servicios Inform?ticos S.L. Tel. 966 160 600 / Fax. 966 160 601

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---