Re: [asterisk-users] ST2030 replacement
Le 08/01/2016 09:18, Glenn Geller (VDOPh) a écrit : Also try vtech vsp725 Thanks, *Glenn* It seems that in France, it's this model that is distributed: "Alcatel Temporis IP300". It looks the same. This model particularly interests me. 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] ST2030 replacement
Le 08/01/2016 00:00, Frank a écrit : Yealink T26P This model is too expansive for me. 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] ST2030 replacement
Le 08/01/2016 08:46, Markos Vakondios a écrit : Grandstream GXP-1628 Are keys of good quality ? 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
[asterisk-users] ST2030 replacement
Hello, I am looking for a replacement for my Thomson ST2030SIP. My specifications are as follows : - 2 lines. - 6 BLF keys. - PoE. Can you give me a return on the models you use ? 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
[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] Dahdi gains
Le 03/09/2013 07:22, Tzafrir Cohen a écrit : See http://lists.digium.com/pipermail/asterisk-dev/2013-August/062219.html I never had read this article before. It answers all my questions. When I have some time, I will test both options to compare their effects on sound quality. 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
[asterisk-users] Dahdi gains
Hi, I'm trying to find the differences between the two CLI gain parameters of Dahdi : dahdi set swgain and dahdi set hwgain. When I change one of these parameters the output of : asterisk -rx dahdi show channel X | grep Gains don't show me any changes. Did dahdi show channel X shows HW or SW Gains ? Whan I set rxgain and txgain in my chan_dahdi.conf file, is it a HW or SW gain ? Thanks in advance, 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] no voicemail on pstn line
Le 26/03/2010 15:01, Landy Landy a écrit : Hello List. I am having problems retreiving voicemails on my system. I noticed when someone leaves a message through the pstn line I can't hear anything. I tested leaving a message from one of the extensions and that can be heard. I don't know if is the type of card I'm using for analog ( cheap X100p modem ) calls but, can't hear any message coming in from that line. Here's voicmail.conf: [general] ; Choose a format to save voicemails as format=gsm volgain=1.1 Hello, I've got problems with volgain and TDM400 card on my voicemail. You can try without it. -- _ -- 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] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- 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] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | | Yeah, well, that's all fine and dandy as long as more capacity is an | option. Many people are already subscribed to the most capacity | available to them and using it. | | b. Apparently man people don't understand that those QoS settings on routers mean little most of the time. Most providers resell QoS as a premium service, so while many waste their time painting their packets those markings get stripped. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAc5R4OeOV2sx4+mAQKEpQ//dYu+9MFaHgHzbBntTMbUHuY4usW5Aq+L crMlq3nYqgi8kWfVShhEozKHvtaYc7J7YBSkE2QprhM/YTp+wE3Oy9NM5GU6Ckhz IDaFNteO62zyxg5ljE81iIQd0tTJjutIf3FQVZBegzpINGIiEkjKBfbx/4UiO6HL bexoS3pnV4xjjS8xO8rMNl8+1XVubpG42K1/alw0G7y7W9Pog+u67+dLx1Tnx0EX RTlAeLZ64u5hy7CXeRdLSM3Onn8IuCnOIP2Py4OEUjLH8K4yMb83IVlhv+KSp4q4 5Tw7LWFsM/NZ0J6xz3MeUnXJHOkNK6Z5UJAfV1LmjiWdpxDCfYDifu6Y5D425+po gd/zHRI+SZJAhzN4l0oWIxSRQdCL6APyFqYFftO9bxAzDoK6EMXADIPvc3Ovb/A0 eUh6rZAe3y5/FfQy29GN23u5//ahFDCzQ9YqhbDjLEc/Z+PLi/lsEdWwWMrUMyus Q4nBs9osuxjRZYWEKUTLal+ItNL/BSiqHurN1T/l3W1/xigYiZHByxEBI2/+jYX6 66wQU6CSE2YC+n9R+rbsAP5OawOTxpXnDdTXEydHCPgdOAS5HmrwTp0t5MNZ4V/N iSGIBBAcV0HJIKRKaeGweIRGStAQPXbfQ9Qha7uYOqnyYwPbt18/vw08YlbdXXCO woCJ+I+AchI= =+Lgr -END PGP SIGNATURE- ___ -- 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
