Re: [asterisk-users] early media (video)
Fronc Hias wrote: Hi! sorry to poke in... but i haven't heard anything since posting my logs :( No real additional thoughts. Everything looks as though it should work. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
[asterisk-users] Voicemail message to text
HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 enable DTLS
On Mon, May 19, 2014 at 11:58 PM, bhavik patel wrote: > Hi All, > > Currently i am integrating webRTC demo. > > I have issue using firefox,someone suggest me to enable DTLS for webRTC > working in firefox using Asterisk. > > I am using Asterisk 11.9.0. > > https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J > > Can any one tell me how to enable DTLS ? I haven't setup a WebRTC environment with DTLS, but you can find a section in the sip.conf.sample file starting with "; DTLS-SRTP CONFIGURATION" that has descriptions of all of the DTLS options. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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
Re: [asterisk-users] Voicemail message to text
On 20 May 2014, at 15:35, Ishfaq Malik wrote: > I was wondering if anyone has implemented voicemail to text and if so, what > package is being used to do so? With the huge variety of different accents and intonations in human speech (even in one country), my experience of all speech-to-text engines has been one of poor accuracy at best. If you need messages-to-text, generally best to use a virtual PA company or similar - at least in my experience. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] autoservice.c MAX_AUTOMONS
Hello, I am currently load testing some new hardware and have been receiving the following warning. Does anyone happen to know if there are any risks or performance implications for increasing the MAX_AUTOMONS value? The current value is 1500. asterisk[30322]: WARNING[30423]: autoservice.c:110 in autoservice_run: Exceeded maximum number of automatic monitoring events. Fix autoservice.c Steven Wheeler -- _ -- 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 enable DTLS
Hi Rusty There are many posts on this list regarding Firefox interoperability. Its not determined if problem is with Firefox or Asterisk. Can someone share working configuration on this list? This will help many users to configure Asterisk to support DTLS-SRTP. *Thanks & Regards,* Amit Patkar On 5/20/2014 5:49 PM, Rusty Newton wrote: On Mon, May 19, 2014 at 11:58 PM, bhavik patel wrote: Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? I haven't setup a WebRTC environment with DTLS, but you can find a section in the sip.conf.sample file starting with "; DTLS-SRTP CONFIGURATION" that has descriptions of all of the DTLS options. -- _ -- 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] authoritative sql definitions for Asterisk Realtime Architecture ARA
AFAIK, there's few columns that are required, and for example sippeers can have as many columns as the corresponding configuration values of peers, and if not present they will claim default values... On 9 May 2014 22:32, Stephen More wrote: > I am trying to find where the authoritative sql definitions for Asterisk > Realtime Architecture ARA are located. I have found many locations but each > and everyone seems to be different. > > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html > > http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example > > Files included with the distribution: > > asterisk-11.9.0/contrib/realtime/mysql/iaxfriends.sql:CREATE TABLE > `iaxfriends` ( > asterisk-11.9.0/contrib/realtime/mysql/meetme.sql:CREATE TABLE meetme ( > asterisk-11.9.0/contrib/realtime/mysql/musiconhold.sql:CREATE TABLE > musiconhold ( > asterisk-11.9.0/contrib/realtime/mysql/queue_log.sql:CREATE TABLE > queue_log ( > asterisk-11.9.0/contrib/realtime/mysql/sippeers.sql:CREATE TABLE IF NOT > EXISTS `sippeers` ( > asterisk-11.9.0/contrib/realtime/mysql/voicemail.sql:CREATE TABLE > voicemail ( > asterisk-11.9.0/contrib/realtime/mysql/voicemail_data.sql:CREATE TABLE > voicemail_data ( > asterisk-11.9.0/contrib/realtime/mysql/voicemail_messages.sql:CREATE TABLE > voicemail_messages ( > > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE > extensions_conf ( > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE cdr ( > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE > sip_conf ( > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE > voicemail_users ( > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE > queue_table ( > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE > queue_member_table > asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE > "queue_log" ( > > Any help is appreciated. > -Stephen More > > -- > _ > -- 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
