Re: [asterisk-users] s/n ratio detection etc...
On Wed, Nov 30, 2011 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN *Sent:* Wednesday, November 30, 2011 8:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] s/n ratio detection etc... ** ** ** ** On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN *Sent:* Wednesday, November 30, 2011 6:25 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] s/n ratio detection etc... Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?...** ** Either I need to finish my coffee or this should be worded better: Sorry about this. This request just came in from a client and we need an answer very quickly. Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? This depends on 1. How are the calls delivered to Asterisk (we will ignore the “other call center” since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? DAHDI 2. What version of Asterisk? 1.8.7 3. Do you want “built-in” methods or could other methods such as daemons be used? either way would be ok. Your best bet as I understand it would be to use dahdi_tools to monitor your lines or to use mixmonitor to record the calls so you can review and tune problems as needed. Either of these options would cost you some overhead in processor usage and disk space. Again, thank you for your help... Much appreciated... Thank you for your quick response -- 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
[asterisk-users] s/n ratio detection etc...
Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... -- _ -- 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] s/n ratio detection etc...
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN Sent: Wednesday, November 30, 2011 6:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] s/n ratio detection etc... Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... Either I need to finish my coffee or this should be worded better: Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? This depends on 1. How are the calls delivered to Asterisk (we will ignore the other call center since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? 2. What version of Asterisk? 3. Do you want built-in methods or could other methods such as daemons be used? -- _ -- 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] s/n ratio detection etc...
On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN *Sent:* Wednesday, November 30, 2011 6:25 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] s/n ratio detection etc... ** ** Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?...** ** ** ** Either I need to finish my coffee or this should be worded better: Sorry about this. This request just came in from a client and we need an answer very quickly. Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? ** ** This depends on **1. **How are the calls delivered to Asterisk (we will ignore the “other call center” since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? DAHDI **2. **What version of Asterisk? 1.8.7 3. Do you want “built-in” methods or could other methods such as daemons be used? either way would be ok. Thank you for your quick response -- 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] s/n ratio detection etc...
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN Sent: Wednesday, November 30, 2011 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] s/n ratio detection etc... On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN Sent: Wednesday, November 30, 2011 6:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] s/n ratio detection etc... Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... Either I need to finish my coffee or this should be worded better: Sorry about this. This request just came in from a client and we need an answer very quickly. Is there a way the detect the individual and cumulative signal-to-noise ratio values for incoming calls to Asterisk (or any other Call Center solution)? This depends on 1. How are the calls delivered to Asterisk (we will ignore the other call center since this is an Asterisk discussion board)? SIP/DAHDI(PSTN/PRI/E1/ETC)? DAHDI 2. What version of Asterisk? 1.8.7 3. Do you want built-in methods or could other methods such as daemons be used? either way would be ok. Your best bet as I understand it would be to use dahdi_tools to monitor your lines or to use mixmonitor to record the calls so you can review and tune problems as needed. Either of these options would cost you some overhead in processor usage and disk space. Thank you for your quick response -- 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