Re: [asterisk-users] Outgoing calls fail in chan_gtalk
On 8/20/2011 12:46 PM, Paul Belanger wrote: Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound google voice calls? confirmed on asterisk 1.8.6.0-rc1 pre-patch behavior: ring-no-answer post-patch behavior: expected -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Sytem Commands not executing
You don't need the path to the php executable if you use hash tags in your script #!/usr/bin/php -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Messina Sent: Saturday, August 20, 2011 10:36 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sytem Commands not executing On 08/20/2011 07:00 AM, Tim King wrote: exten = h,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php do you need the -f option to php? exten = h,n,System(/usr/bin/php -f /var/lib/asterisk/bin/faxnotify.php -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -- _ -- 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] Outgoing calls fail in chan_gtalk
On 11-08-21 02:54 AM, Jeremy Kister wrote: On 8/20/2011 12:46 PM, Paul Belanger wrote: Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound google voice calls? confirmed on asterisk 1.8.6.0-rc1 pre-patch behavior: ring-no-answer post-patch behavior: expected Thanks everybody, I've merged the patch. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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
[asterisk-users] Flite module for asterisk
Version 2.0 of app_flite just got released. Flite For Asterisk provides the Flite dialplan application, which allows you to use the Flite TTS Engine with Asterisk. It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with asterisk 1.6 , 1.8 , 10. http://zaf.github.com/Asterisk-Flite/ Lefteris Zafiris -- _ -- 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] espeak module for asterisk
Version 2.0 of app_espeak just got released. eSpeak For Asterisk provides the Espeak dialplan application, which allows you to use the Espeak speech synthesizer with Asterisk. It supports the following languages: Afrikaans, Albanian, Armenian,Cantonese, Catalan, Croatian, Czech, Danish, Dutch, English, Esperanto, Estonian, Finnish, French, Georgian, German, Greek, Hindi, Hungarian, Icelandic, Indonesian, Italian, Kannada, Kurdish, Latvian, Lojban, Macedonian, Malayalam, Mandarin, Norwegian, Polish, Portuguese, Romanian, Russian, Serbian, Slovak, Spanish, Swahili, Swedish, Tamil, Turkish, Vietnamese, Welsh. It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. Works with asterisk 1.6 , 1.8 , 10. http://zaf.github.com/Asterisk-eSpeak/ Lefteris Zafiris -- _ -- 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] How is a ping test delay ms different from status in Asterisk sip show peers?
On 08/20/2011 02:24 PM, Bruce B wrote: What's the point of having the metrics then? They are inaccurate and deceiving. If there is no benefit to showing the real metrics then why not change it to Status = Reachable than showing a number? Because it's still more useful than not having it? If I see someone with an Asterisk RTT of ~200 ms in 'sip show peers', I know their phone is working fine. But if I see 3000 ms, they are probably lagged due to bandwidth contention or other problem. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] How to get presence using AMI
If the peers are SIP you could do: akl*CLI manager show command SIPpeers Action: SIPpeers Synopsis: List SIP peers (text format) Privilege: system,all Description: Lists SIP peers in text format with details on current status. Variables: ActionID: idAction ID for this transaction. Will be returned. Basically you should connect to Asterisk, go into the console (asterisk -r) and then type manager show commands. Read through them, learn what they all do and you'll pretty quickly get a feel for what you can do and how to do it. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] allow anonymous call
Hello, How to allow inbound anonymous call on asterisk ? Sincerely, Tseveen -- _ -- 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] allow anonymous call
On 08/22/2011 01:38 AM, tseveendorj wrote: How to allow inbound anonymous call on asterisk ? allowguest = yes, in sip.conf [general] section. However, I do not advise it. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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
