[asterisk-users] TE420, is it possible do disable span (red blinking)?
Hello! Just got new server with TE420. Not all four spans will be used immediately, but spans not configured or not connected blink red light. Is it possible to turn span off, so my colleagues will not eventally tell me that something is wrong with asterisk? :-) Thank you! -- _ -- 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] TE420, is it possible do disable span (red blinking)?
Just dont configure those spans n related channels inside chan_dahdi.conf Mitul On Nov 1, 2013 3:38 PM, Dmitry Melekhov d...@belkam.com wrote: Hello! Just got new server with TE420. Not all four spans will be used immediately, but spans not configured or not connected blink red light. Is it possible to turn span off, so my colleagues will not eventally tell me that something is wrong with asterisk? :-) Thank you! -- __**__**_ -- 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-**usershttp://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] dahdi fax catch-22 [SOLVED]
On Wed, 2013-10-30 at 16:17 -0600, Greg Woods wrote: On Wed, 2013-10-30 at 11:12 -0500, Richard Mudgett wrote: One thing you must remember about the [channels] section in chan_dahdi.conf is that *all* configuration in the section is cumulative. For example: [channels] echocancel = yes faxdetect=incoming signaling=fxs_ks channel=1 ; Channel 1 is created at this point with echocancel enabled, faxdetect is incoming, ; and fxs_ks signaling. faxdetect=no channel=2 ; Channel 2 is created at this point with echocancel enabled, faxdetect is no, Thank you very much, indeed I did not know that. So I tried it, specifying faxdetect=incoming only for the last (POTS) channel. It appears to have sent an outgoing fax correctly ...and just to close this thread, the incoming faxes are also working correctly. Thanks very much for the configuration tip. --Greg -- _ -- 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
[asterisk-users] Redirect a GSM call through Wifi to a SIP phone
Hi all, I've try to search Google about this without any chance. I want to know if it's possible to use a mobile phone application for redirect automatically incoming calls of a GSM phone connected to Wifi network to a Sip phone. I've try to use different mobile phones SIP clients without any success. No one of them can redirect calls automatically. I've got Android and BlackBerry phones. Thanks. Sil -- _ -- 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] Redirect a GSM call through Wifi to a SIP phone
On Friday 01 November 2013, Sil wrote: I want to know if it's possible to use a mobile phone application for redirect automatically incoming calls of a GSM phone connected to Wifi network to a Sip phone. I've try to use different mobile phones SIP clients without any success. No one of them can redirect calls automatically. I've got Android and BlackBerry phones. What you want is not just a SIP client; it also has to integrate with the phone's own GSM stack. You probably will have the most success with Android, because you are going to need well-documented Source Code to stand a chance of getting anywhere. You will need an Open Source SIP client and the Source Code for the stock Android GSM telephony app. You then will have to bodge the two together somehow . Actually, as long as the data packets coming from Android GSM phone are already compressed using the GSM codec, you shouldn't have to do anything to them to re-send them out over SIP; and similarly for packets arriving by the SIP connection to re-send over GSM. This really is not going to be a trivial project. Good luck with it -- you're going to need it :) -- AJS Answers come *after* questions. -- _ -- 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] Realtime Call Files
On 10/31/13, 8:44 AM, Rizwan Hisham wrote: Hi all, Is there any way of originating calls in future without using call files? We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other complications. Any crazy ideas would be helpful. Thanks The easiest way to do this would be with AMI and originate your calls in realtime. That way you do not have to worry about which server will handle the call, the one you connect to will. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] Redirect a GSM call through Wifi to a SIP phone
Hi, On 11/1/2013 5:02 PM, A J Stiles wrote: You probably will have the most success with Android, because you are going to need well-documented Source Code to stand a chance of getting anywhere. You will need an Open Source SIP client and the Source Code for the stock Android GSM telephony app. You then will have to bodge the two together somehow . please correct me if I'm wrong, but if the above was possible, then also, current SIP softphone apps would be able to do - conference calls with a mixture of GSM and SIP endpoints - transfer calls from/to GSM to/from SIP which as far as I know, none of the apps offer. If I remember correctly, one of the softphone app programmers provided a reason why the above will never work: you can't access a GSM call from an app due to security reasons. I'm not sure what percentage I'm right, but I'd strongly suggest a GSM gateway for the function either with ethernet and SIP support, or an FXO port. Much easier project. regards Adam -- _ -- 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
[asterisk-users] Redirect a GSM call through Wifi to a SIPphone
Any sip softphone will work. Linphone is free. I have tested many . All works well with audio. -- _ -- 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
