Re: [asterisk-users] dial in group
As SIP is not Analog FXO, my comments do not apply to them. I have no idea if or which analog adapters might detect line voltage or dialtone. Paul wrote: > Do the SIP-FXO gateway devices do any better? > > Eric "ManxPower" Wieling wrote: > >> Asterisk does not detect analog ports with no line plugged in. It does >> not test for dialtone before dialing (this applies to all analog cards >> except the X100P). >> >> Rilawich Ango wrote: >> >> >>> It works if it specified the port exactly plugged to PSTN. I want to >>> clarify the dial command here. >>> >>> Dial(zap/g1/1234567) >>> >>> It will try channel 1, if it is busy, congested then it will try >>> channel 2 and so on, right? >>> I wonder if I don't plug the PSTN to channel 1, there should not be a >>> dial tone on it. Why it still try channel 1 and make call using it? >>> >>> On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >>> >>> On Sat, 24 Nov 2007, Rilawich Ango wrote: > I have a TDM400 with all FXO module in it. Only one channel (say > channel 3) is plugged to PSTN. In my understand, a dial command > Dial(zap/g1/12345677) should search an available channel, which is 3, > in group 1 to make a call. However, I found that it will still use > channel 1 to make call even it hasn't plugged to the PSTN. Below are > the conf files. > > --zapata.conf-- > group=1 > signalling=fxs_ks > context=incoming > channel => 1-8 > > You really only want channel => 3 > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
Do the SIP-FXO gateway devices do any better? Eric "ManxPower" Wieling wrote: >Asterisk does not detect analog ports with no line plugged in. It does >not test for dialtone before dialing (this applies to all analog cards >except the X100P). > >Rilawich Ango wrote: > > >>It works if it specified the port exactly plugged to PSTN. I want to >>clarify the dial command here. >> >>Dial(zap/g1/1234567) >> >>It will try channel 1, if it is busy, congested then it will try >>channel 2 and so on, right? >>I wonder if I don't plug the PSTN to channel 1, there should not be a >>dial tone on it. Why it still try channel 1 and make call using it? >> >>On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >> >> >>>On Sat, 24 Nov 2007, Rilawich Ango wrote: >>> >>> >>> I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel => 1-8 >>>You really only want >>> >>> channel => 3 >>> >>> ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
Asterisk does not detect analog ports with no line plugged in. It does not test for dialtone before dialing (this applies to all analog cards except the X100P). Rilawich Ango wrote: > It works if it specified the port exactly plugged to PSTN. I want to > clarify the dial command here. > > Dial(zap/g1/1234567) > > It will try channel 1, if it is busy, congested then it will try > channel 2 and so on, right? > I wonder if I don't plug the PSTN to channel 1, there should not be a > dial tone on it. Why it still try channel 1 and make call using it? > > On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >> On Sat, 24 Nov 2007, Rilawich Ango wrote: >> >>> I have a TDM400 with all FXO module in it. Only one channel (say >>> channel 3) is plugged to PSTN. In my understand, a dial command >>> Dial(zap/g1/12345677) should search an available channel, which is 3, >>> in group 1 to make a call. However, I found that it will still use >>> channel 1 to make call even it hasn't plugged to the PSTN. Below are >>> the conf files. >>> >>> --zapata.conf-- >>> group=1 >>> signalling=fxs_ks >>> context=incoming >>> channel => 1-8 >> You really only want >> >>channel => 3 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
On Sun, 25 Nov 2007, Rilawich Ango wrote: > It works if it specified the port exactly plugged to PSTN. I want to > clarify the dial command here. > > Dial(zap/g1/1234567) > > It will try channel 1, if it is busy, congested then it will try > channel 2 and so on, right? Yes. > I wonder if I don't plug the PSTN to channel 1, there should not be a > dial tone on it. Why it still try channel 1 and make call using it? Because asterisk can't tell if an analogue line is plugged in or not. To get a dial-tone, it would have to activate the line (ie. "lift the handset") and it's not going to do that. It's relying on the channel instruction in the zapata.conf file to tell it what lines are really live, so get them right and everything else will "just work". Gordon > On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >> >> On Sat, 24 Nov 2007, Rilawich Ango wrote: >> >>> I have a TDM400 with all FXO module in it. Only one channel (say >>> channel 3) is plugged to PSTN. In my understand, a dial command >>> Dial(zap/g1/12345677) should search an available channel, which is 3, >>> in group 1 to make a call. However, I found that it will still use >>> channel 1 to make call even it hasn't plugged to the PSTN. Below are >>> the conf files. >>> >>> --zapata.conf-- >>> group=1 >>> signalling=fxs_ks >>> context=incoming >>> channel => 1-8 >> >> You really only want >> >>channel => 3 >> >> here if it's only channel 3 that's plugged in. >> >> Gordon >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
It works if it specified the port exactly plugged to PSTN. I want to clarify the dial command here. Dial(zap/g1/1234567) It will try channel 1, if it is busy, congested then it will try channel 2 and so on, right? I wonder if I don't plug the PSTN to channel 1, there should not be a dial tone on it. Why it still try channel 1 and make call using it? On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: > > On Sat, 24 Nov 2007, Rilawich Ango wrote: > > > I have a TDM400 with all FXO module in it. Only one channel (say > > channel 3) is plugged to PSTN. In my understand, a dial command > > Dial(zap/g1/12345677) should search an available channel, which is 3, > > in group 1 to make a call. However, I found that it will still use > > channel 1 to make call even it hasn't plugged to the PSTN. Below are > > the conf files. > > > > --zapata.conf-- > > group=1 > > signalling=fxs_ks > > context=incoming > > channel => 1-8 > > You really only want > >channel => 3 > > here if it's only channel 3 that's plugged in. > > Gordon > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
On Sat, 24 Nov 2007, Rilawich Ango wrote: > I have a TDM400 with all FXO module in it. Only one channel (say > channel 3) is plugged to PSTN. In my understand, a dial command > Dial(zap/g1/12345677) should search an available channel, which is 3, > in group 1 to make a call. However, I found that it will still use > channel 1 to make call even it hasn't plugged to the PSTN. Below are > the conf files. > > --zapata.conf-- > group=1 > signalling=fxs_ks > context=incoming > channel => 1-8 You really only want channel => 3 here if it's only channel 3 that's plugged in. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial in group
I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel => 1-8 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
