[asterisk-users] How to update sound files?
Hi. I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk, but it still plays old ones. I've removed old files completely then, but still no effect. I've searched for some cahce files, but didn't find any of them. So, could you tell me please, what is an appropriate way to update my sound files? :) Thanks. Asterisk 1.6.1.20 -- _ -- 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 update sound files?
what kind of update have you done? which is your goal? DTMF commands of our IVR have been modified, so I'd like to replace old wav-files with new ones to reflect changes. Probably if nothing changes is because the files you need to update aren't in the above path (i.e. if you want to change language) Well, I'm sure the path is correct. Il 28/01/2011 11:27, Сикорский Сергей ha scritto: Hi. I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk, but it still plays old ones. I've removed old files completely then, but still no effect. I've searched for some cahce files, but didn't find any of them. So, could you tell me please, what is an appropriate way to update my sound files? :) Thanks. Asterisk 1.6.1.20 -- _ -- 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 -- _ -- 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 update sound files?
I found the problem. It seemed to be something stupid and it really was. The name of a new file had missing character. Sorry for this thread :) 28.01.2011 15:15, Сикорский Сергей пишет: what kind of update have you done? which is your goal? DTMF commands of our IVR have been modified, so I'd like to replace old wav-files with new ones to reflect changes. Probably if nothing changes is because the files you need to update aren't in the above path (i.e. if you want to change language) Well, I'm sure the path is correct. Il 28/01/2011 11:27, Сикорский Сергей ha scritto: Hi. I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk, but it still plays old ones. I've removed old files completely then, but still no effect. I've searched for some cahce files, but didn't find any of them. So, could you tell me please, what is an appropriate way to update my sound files? :) Thanks. Asterisk 1.6.1.20 -- _ -- 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 -- _ -- 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 -- Sergej Sikorsky, Head of IT Department of Lanet Network LTD Tel.: +38 096 29-79-299 www.lanet.ua -- _ -- 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] Wise selecting of outgoing channel
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ? ?? Sent: Wednesday, December 22, 2010 4:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Wise selecting of outgoing channel Hi. We have 3 channel for _outgoing_ calls and would like to use them equally (by turn). Is there any 'queue' for outgoing calls? Now I have to select the second channel manually when the first is busy and so on. Does Asterisk have any functionality to do it automatically? That's what the G and R functions of dial are for. Dial(DAHDI/G1) selects the 3 lines in reverse order (3,2,1). Dial(DAHDI/g1) selects them in ascending order (123). Dial(DAHDI/r1) is round-robin and Dial(DAHDI/R1) is round-robin-reverse. This is defined in the Asterisk Guide. Yes, that's how it works for PRI. But what about three _different_ independent SIP channels to GSM-gateways? _ -- 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 -- Sergej Sikorsky, Head of IT Department of Lanet Network LTD Tel.: +38 096 29-79-299 www.lanet.ua -- _ -- 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] How to read core-en_US.xml
Hi. There is /doc/core-en_US.xml in asterisk 1.8 source tree. Is this file generated from documentation comments of apps/app_*.c files? And how this file can be used? How can I convert it to pdf/html in order to use it as applications documentation? -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? Thanks, --Warren Selby On Oct 14, 2010, at 11:36 PM, Matt Darnellmattdarn...@gmail.com wrote: Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selbywcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnellmattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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 -- _ -- 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
