[Asterisk-Users] No dial tone

2005-01-12 Thread ismaelg
Hello all, My problem is that when I call from an extension to another, I ear the dial tones, but when I make a call using the Zap or Capi channels I do not ear the dial tones. Why this could be happen? Any clue will be appreciated. Thanks. Ismael gil.

[Asterisk-Users] Zaptel config

2005-01-11 Thread ismaelg
Hello all, I am having a lot of problems with zaptel channels, I have got an TDM02B, and I don't know how setup /etc/zaptel.con and /etc/asterisk/zapata.conf for use it on asterisk. Some one could help me with this configuración? My problem is about the type of signalling Thanks, Regards. Ismael

[Asterisk-Users] TDM box Hardware

2005-01-11 Thread ismaelg
Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details?

[Asterisk-Users] Zaptel problems

2005-01-10 Thread ismaelg
Hello all, I recently install a TDM04B with only 2 FXS modules. I just install Asterisk 1.0.3, zaptel 1.0.3, and libpri 1.0.3, all of these stable packages. I configured /etc/zaptel.conf like the following loadzone = us #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de

Re: [Asterisk-Users] More Zaptel problems

2005-01-10 Thread ismaelg
. But when I try to start asterisk, I get a box crash just after parsing musiconhold. Any clue? Thanks Ismael Gil. ismaelg wrote: Hello all, I recently install a TDM04B with only 2 FXS modules. I just install Asterisk 1.0.3, zaptel 1.0.3, and libpri 1.0.3, all of these stable packages. I

Re: [Asterisk-Users] More Zaptel problems

2005-01-10 Thread ismaelg
It crashes, here the last messages asterisk wrote in stdout before hangs. [app_hasnewvoicemail.so] = (Indicator for whether a voice mailbox has messages in a given folder.) == Registered application 'HasVoicemail' == Registered application 'HasNewVoicemail' [format_wav_gsm.so] = (Microsoft

[Asterisk-Users] Reading mysql sip friends

2004-12-13 Thread ismaelg
Hello, I am trying to setup an asterisk to store users datails in a mysql database. Explained here http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers All works well, I create a user in the database, but asterisk seems that can't read the data, I create a mysql user with enough

[Asterisk-Users] Mysql configuration interface

2004-12-10 Thread ismaelg
Hello, I trying to configure asterisk to store sip and iax2 user in a mysql database. All goes well, but my problem is when i try to add a new user (sip or iax). I have look for an aplication with a web interface that lets us manage the user account in asterisk without success . How could I

[Asterisk-Users] A waning console error

2004-12-09 Thread ismaelg
Hello, I am getting this kind of Warning in the Asterisk console, but i don't know why. WARNING[8200]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Could you give some clue to solve this problem? Thanks in advice. Ismael.

[Asterisk-Users] Forwarding calls

2004-11-23 Thread ismaelg
Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I

[Asterisk-Users] edirecting calls with Asterisk

2004-11-22 Thread ismaelg
Hello, I am trying a couple of days before to set up asterisk to redirects an incoming call if the extension dialed is busy without success. I just try to use 'Gotoif' command, with bad luck, it can't do what i want. Anybody could helpme? ani clue will be appreciated. Regards. Ismael.

[Asterisk-Users] Asterisk DNS issue

2004-11-11 Thread ismaelg
Hello all, I just configure Bind 9 in our LAN to resolve the Asterisk name sip.bussines.com for our phones. I want that when a local extensión calls to another local extension, the phone shows Extension@DNS name instead of Extension@ip address like now happens. In all my phones I configure

[Asterisk-Users] I can't solve my problems with the IVR

2004-10-19 Thread ismaelg
Hello all, I'm still having problems with the IVR options. When I press on my mobile phone one of the digits related in the IVR options, press 1 for .,press 2 for.., press 3 for.. After I press the one, the second or the tirth key on my mobile phone, I can't hear nothing more, I can't

[Asterisk-Users] IVR option problem

2004-10-18 Thread ismaelg
Hello all, I'm trying setting up an IVR on a Asterisk Soho PBX. My problem is when I dial the IVR extensin from an Asterisk internal extension all goes well, but when I dial the external number of the IVR, e.g. 119235656, the PSTN number of my asterisk, I get the same IVR menu but when I

[Asterisk-Users] A question with voice Menu

2004-10-13 Thread ismaelg
Hello, I'm having the following problem in my asterisk config. I have a little voice menu, with two options, The welcome message looks like that, 1- press 1, to dial an extension 2- press 2, to speak with an operator. If I press 1, I get the following message Dial the extensión number

[Asterisk-Users] Changing the default language

2004-10-13 Thread ismaelg
Hello all, I am tring to change the default language in Asterisk, exactly for the Voicemail messages. I trying with the option Language=fr in the voicemail.conf global section, without success. I trying with the Setlanguage(fr) in the extensions.conf global section, but without success too.

[Asterisk-Users] Problems with voice menu

2004-10-11 Thread ismaelg
Hello all, I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten =

Re: [Asterisk-Users] Problems with voice menu

2004-10-11 Thread ismaelg
Thank you Christopher, Imade the changes you told me, but, when I try to make an incoming call, in the Asterisk console, I get -- Hungup 'IAX2/[EMAIL PROTECTED]:4569/9' -- Executing Dial("SIP/aurelio-92fe", "IAX2/501050:[EMAIL PROTECTED]/501050|60|r") in new stack -- Called