Hi, I have a very newbie question, I asked this at the digium support, but they did not answered yet, so I hope you can help. I just bought the digium developer kit (TDM400P), installed everything, all messages seems ok, but I can't make it work. * a normal telephone and a line connected to the board * modprobe wcfxs = ok leds go on an I hear a small click at the phone. * ztcfg = ok (two channels found ) Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) * asterisk running with all verbose in console and remote mode. only chan_skinny and chan_phone had problems to load. * The config files seems ok. Problem: According to the config files, by pressing 9 I should have the line to dial outside, but absolutely nothing happens. I tried the other numbers, with no response, The phone is mute (only a subliminar energy sound and the dtmf sound when I press the keys). How can I find what I did wrong? the verbose and the messages at /var/log/messages says everything is ok. When it was compiling, there was a little issue at the zaptel drivers, and I had to change a inline function position to compile correctly. (source from cvs) I'm in brazil, can this be some problem with the telephone line? I changed the setting to use loadzone and defaultzone "br", but this not worked either. Can I execute something at the CLI to hear it at the phone, just to see if everything is ok? []s, gandhi -- Ricardo Andere de Mello Presidente do Quilombo Digital 55 11 3271-7928
a small correction, chan_skinny and *chan_oss* failed []s, gandhi Ricardo Andere de Mello wrote:> Hi, > I have a very newbie question, I asked this at the digium support, but > they did not answered yet, so I hope you can help. > > I just bought the digium developer kit (TDM400P), installed > everything, all messages seems ok, but I can't make it work. > > * a normal telephone and a line connected to the board > * modprobe wcfxs = ok > leds go on an I hear a small click at the phone. > * ztcfg = ok (two channels found ) > Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: > FXS Kewlstart (Default) (Slaves: 04) > * asterisk running with all verbose in console and remote mode. > only chan_skinny and chan_phone had problems to load. > * The config files seems ok. > > Problem: > According to the config files, by pressing 9 I should have the line to > dial outside, but absolutely nothing happens. I tried the other > numbers, with no response, The phone is mute (only a subliminar energy > sound and the dtmf sound when I press the keys). > > How can I find what I did wrong? the verbose and the messages at > /var/log/messages says everything is ok. > > When it was compiling, there was a little issue at the zaptel drivers, > and I had to change a inline function position to compile correctly. > (source from cvs) > > I'm in brazil, can this be some problem with the telephone line? I > changed the setting to use loadzone and defaultzone "br", but this not > worked either. > > Can I execute something at the CLI to hear it at the phone, just to > see if everything is ok? > > []s, gandhi >-- Ricardo Andere de Mello Presidente do Quilombo Digital 55 11 3271-7928
Two items to consider .... 1) make sure you have the connections right ... Your FXO port on the card (the one you will signal using FXS) is connected to the incoming phone line ... Your FXS port on the card (the one you will signal using FXO) is connected to the analog telephone you want to use ... If you have the card installed properly and the telephone handset connected to the correct port, you should get dial tone as soon as you pick up the hook ... 2) the "sample" dial plan that comes with Asterisk should not be expected to run as supplied ... It is just an example of what is possible ... If it actually works as provided, you got lucky ... You will need to configure a dial plan that is appropriate for your planned use ... Dial 9, is really optional ... You can just as easily define a dial 8 system if you want to do so ... I use dial 9 for PSTN, dial 8 for SIP VoIP provider and dial 7 for IAX provider ... A simpler system with only PSTN access might not use any dial prefix ... Or if you have designed a least cost routing table that looks at the number dialed and automatically decides what service to use, you would not use any dial prefix ... Unless you want to give the caller the option of choosing an outbound path ... There are many options to consider ... Best thing to do as you start out is get something simple working ... Get it to dial out, ring on inbound and accept voicemail ... Once you have it doing the basics, you will have learned enough getting that far along that some of the more advanced features will not be so intimidating to tackle ... There are many examples of dial plans avaialble for review on the wiki at www.voip-info.org ... This is an excellent place to get some ideas on how to setup a dial plan that will do what you want to do ... Regards G.Hendershot
thanks a lot for your help, I will try to make the dialplan. []s, gandhi Gary G. Hendershot wrote:>Two items to consider .... > >1) make sure you have the connections right ... Your FXO port on the card >(the one you will signal using FXS) is connected to the incoming phone line >.... Your FXS port on the card (the one you will signal using FXO) is >connected to the analog telephone you want to use ... If you have the card >installed properly and the telephone handset connected to the correct port, >you should get dial tone as soon as you pick up the hook ... > >2) the "sample" dial plan that comes with Asterisk should not be expected to >run as supplied ... It is just an example of what is possible ... If it >actually works as provided, you got lucky ... You will need to configure a >dial plan that is appropriate for your planned use ... > > >Dial 9, is really optional ... You can just as easily define a dial 8 system >if you want to do so ... I use dial 9 for PSTN, dial 8 for SIP VoIP provider >and dial 7 for IAX provider ... A simpler system with only PSTN access might >not use any dial prefix ... Or if you have designed a least cost routing >table that looks at the number dialed and automatically decides what service >to use, you would not use any dial prefix ... Unless you want to give the >caller the option of choosing an outbound path ... There are many options to >consider ... > >Best thing to do as you start out is get something simple working ... Get it >to dial out, ring on inbound and accept voicemail ... Once you have it doing >the basics, you will have learned enough getting that far along that some of >the more advanced features will not be so intimidating to tackle ... > >There are many examples of dial plans avaialble for review on the wiki at >www.voip-info.org ... This is an excellent place to get some ideas on how to >setup a dial plan that will do what you want to do ... > >Regards > >G.Hendershot > > > >-- Ricardo Andere de Mello Presidente do Quilombo Digital 55 11 3271-7928