[asterisk-users] incoming call FXO
Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler I have not this 's' extension. Anybody knows what happen? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)... [Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)... [Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler I have not this 's' extension. Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten = s,1,Verbose( - - - Call received - - - ) exten = s,n,Playback(hello-world) extent = s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
Ok. Problem solved . Thank you very much!!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 15 Sep 2010 09:56:36 -0400 From: zisha...@gmail.com To: kpflem...@digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] incoming call FXO As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten = s,1,Verbose( - - - Call received - - - ) exten = s,n,Playback(hello-world) extent = s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
