[asterisk-users] asterisk 11.14 - voicemail incorrect duration
Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s@macro-voicemail:77]VoiceMail(SIP/XY-0005175a, aNumber,su) in new stack [Jan 26 15:24:04] -- SIP/YX-0005175a Playing 'beep.gsm' (language 'de') [Jan 26 15:24:04] -- x=0, open writing: /var/spool/asterisk/voicemail/default/aNumber/tmp/rVBkAm format: wav, 0x7fd884a33128 [Jan 26 15:24:10] -- User hung up [Jan 26 15:24:10] -- Recording was 2 seconds long but needs to be at least 3 - abandoning So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only count 2. What can be the reason? It is not silence. Sincerely, Dominique -- _ -- 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] PJSIP vs SIP channeltype
Matt Hoskins wrote: Hello, Kia ora, I’m currently evaluating asterisk 13 (Currently on 11). We’re testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? There is not. Cheers, -- 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
Re: [asterisk-users] Dialing from phonebook, and hiding the dialed number from the user.
Hi, does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course. It seems Mitel can do this, and I have a use case where this is a requirement. I don't know about a phone that can do that, but I can give you another possibility that might be an acceptable substitute. You could alias the numbers in the phone so that in Asterisk they do something different. In the phonebook you would have something like: Bob Smith: 1000. Then in Asterisk, you have as part of your dialplan that 1000 would dial Bob Smith's real number. The user of the phone would only ever see the number 1000 associated with Bob Smith. The history would still be there in the phone, but again, it would just show 1000 as well. How far you take this would depend somewhat on how often the underlying numbers change. You could hard code the numbers in your Asterisk dialplan or you could plug them into a database so that they are easier to change in the future. Would that work for what you need?-- _ -- 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] Dialing from phonebook, and hiding the dialed number from the user.
Hi, does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course. It seems Mitel can do this, and I have a use case where this is a requirement. Thanks, Antonio -- _ -- 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] Need help interpreting SDP on failing WebRTC connection
Hi, I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, I got nothing. My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital ocean, and from my laptop I also have a direct VPN connection to the asterisk server my laptop being 192.168.241.10 and asterisk being 192.168.241.30 I do not understand several things: 1. asterisk seems to be telling sipml5 to send audio to it's public ip addres, but * sends to 192.168.241.10 2. the asterisk output shows one way RTP flow. There's no sound from chrome. I am trying to debug, but need some explanation about the SDP with respect to WebRTC and ICE, I hope someone can intersperse the output with comments? Thanks, Antonio Below are the asterisk log, and the Javascript console output: http://pastebin.com/dTFTrzg6 -- _ -- 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] PJSIP vs SIP channeltype
Hello, I'm currently evaluating asterisk 13 (Currently on 11). We're testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -- _ -- 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] Need some help interpreting SDP on a failing WebRTC connection
Hi, I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, I got nothing. My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital ocean, and from my laptop I also have a direct VPN connection to the asterisk server my laptop being 192.168.241.10 and asterisk being 192.168.241.30 I do not understand several things: 1. asterisk seems to be telling sipml5 to send audio to it's public ip addres, but * sends to 192.168.241.10 2. the asterisk output shows one way RTP flow. There's no sound from chrome. I am trying to debug, but need some explanation about the SDP with respect to WebRTC and ICE, I hope someone can intersperse the output with comments? Thanks, Antonio Below are the asterisk log, and the Javascript console output. http://pastebin.com/dTFTrzg6 -- _ -- 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] asterisk 11.14 - voicemail incorrect duration
Hi Stefan, Stefan Tichy asteri...@pi4tel.de schrieb am Mon, 26. Jan 23:56: Hi Dominique On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only count 2. What can be the reason? It is not silence. Are you sure? Yes, im sure. I have looked at the time and talked for at least 4 seconds. In CLI log are 5-6 seconds visible between open to writing and Hang up. Nevertheless, Asterisk writes about two seconds. The value for silencethreshold (140) is unusually large. It would be worth a try to set the value down. In asterisk 1.6 this value was still good. But that is far back again... I will write again. Greetings Dominique -- _ -- 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] customizing Asterisk CLI
Hi Martin, please paste the error or a print screen in cli. On Sun, Jan 25, 2015 at 8:11 AM, Martin Vegter martin.veg...@aol.com wrote: Hello, when I am in the Asterisk CLI, I can exit with 'exit' or 'quit'. Ctrl+d has no effect. Is there any way to bind Ctrl+d to exit/quit ? Also, when I am in asterisk CLI, I can use command history and readline functions such as CTRL+r to search. But not all functions are available. For example, the alternate mappings for page up and page down to search the history do not work. They work in everything else (bash, mysql, ..) $ cat /etc/inputrc \e[5~: history-search-forward \e[6~: history-search-backward is there a way to make it work in asterisk ? I am using Asterisk 11.13 on Debian Wheezy. -- _ -- 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 -- Benjamín Garzón -- _ -- 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] Call Recording doesn't work
Hi all, on my atserisk box call recording and cdr doesn't work. In the log files I have a strange entry - does this have something to do with that? Version: Asterisk 13.1.0 Host: debian wheezy 7.7 Thanks a lot for a brief hint . Walter. *** [2015-Jan-26 11:34:04] [PHP-DEPRECATION_WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your php.ini [2015-Jan-26 11:34:04] [PHP-WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): Unable to load dynamic library '/usr/lib/php5/20100525/digium_register.so' - /usr/lib/php5/20100525/digium_register.so: cannot open shared object file: No such file or directory [2015-Jan-26 11:34:06] [PHP-DEPRECATION_WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your php.ini [2015-Jan-26 11:34:06] [PHP-WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): Unable to load dynamic library '/usr/lib/php5/20100525/digium_register.so' - /usr/lib/php5/20100525/digium_register.so: cannot open shared object file: No such file or directory [2015-Jan-26 11:34:07] [PHP-DEPRECATION_WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your php.ini [2015-Jan-26 11:34:07] [PHP-WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): Unable to load dynamic library '/usr/lib/php5/20100525/digium_register.so' - /usr/lib/php5/20100525/digium_register.so: cannot open shared object file: No such file or directory [2015-Jan-26 11:34:20] [PHP-DEPRECATION_WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your php.ini [2015-Jan-26 11:34:20] [PHP-WARNING] (/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph p:29) - dl(): Unable to load dynamic library '/usr/lib/php5/20100525/digium_register.so' - /usr/lib/php5/20100525/digium_register.so: cannot open shared object file: No such file or directory *** -- _ -- 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] asterisk 11.14 - voicemail incorrect duration
Hi Dominique On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only count 2. What can be the reason? It is not silence. Are you sure? The value for silencethreshold (140) is unusually large. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- 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
