[asterisk-users] choppy sound with playback, background, etc... but not with musiconhold

()... 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[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L

Re: [asterisk-users] 200+ analog phones connected to FXS modules

On nov/30/2006, Vieri wrote: Is there another way of doing this (hopefully cheaper and more convenient)? VoIP Gateways with 48 FXS ports. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601

Re: [asterisk-users] quadbri + kernel 2.6.18.1

On nov/16/2006, Tzafrir Cohen wrote: I'm not sure it was resolved yet. As a workaround, issue 'ztcfg -s' before removing that module. Thanks. I will try. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601

[asterisk-users] quadbri + kernel 2.6.18.1

module? Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] TDM, loopstart and modules GSM Nokia32

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[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601

[asterisk-users] quadbri + tdm400p + modem-fax

to 9600bps = no changes - disable echo cancel in all channels = no changes - Answer() before Dial (to detect fax tone and disable echo cancel) = no changes Somebody has tried a configuration like this? Thanks in advance. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios

Re: [asterisk-users] quadbri + tdm400p + modem-fax

On sep/28/2006, Steve Underwood wrote: Lots have tried it. it doesn't work. With Sangoma cards it will work? Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601

Re: [asterisk-users] 64 analog phones

On sep/22/2006, mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? Maybe a Top-Gate SIP gateway. It supports 16, 24 or 48 FXS ports. You don't need T1 or E1 extra in the Asterisk machine, only one ethernet card. -- Paco

[asterisk-users] modifying the INVITE headers

=rfc2833 [EMAIL PROTECTED] - It's possible to change the INVITE headers to put the correct host? Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth

Re: [asterisk-users] modifying the INVITE headers

fromdomain=telefonica.net dtmfmode=rfc2833 [EMAIL PROTECTED] --- Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] ExtensionState always returns 1

Hello, I have one SIP extension that can make calls, but not receive. If I use ExtensionState in the manager, I always see 1 (In Use), with the softphone not in use. Is there a way to change the extension status? Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux

[asterisk-users] QuadBRI + TDM + GSM hangup problems

=1-2,4-5,7-8,10-11 -- Can someone explain me why the call is terminated? How can I solve this problem? Thanks in advance. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601

[asterisk-users] SOLVED: QuadBRI + TDM + GSM hangup problems

On jul/19/2006, Paco Brufal wrote: The system can receive and make calls perfecty, via ISDN and GSM. But when I configure Asterisk to redirect the calls from ISDN to a mobile telephone (via the GSM modules), when the mobile phone answers, the call is terminated: The solution

[Asterisk-Users] hang up when pickup analog phone

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

[Asterisk-Users] attended call transfer in 1.0.5

attended transfers work? It's a feature really needed :( Thanks in advance :) -- Paco Brufal [EMAIL PROTECTED] Servitux Servicios Informáticos S.L. http://www.servitux.com/ Telefono: 966090600 Fax: 966090601 ___ Asterisk-Users mailing list