Re: [asterisk-users] auto answer
Hello, You could use Answer-After for that. But, afaik there is no definitive description in the RFCs on how it is used. You would have to enable such features on the telephones too. And I would expect that different phone manufacturers would probably use different mechanisms to enable such an option. Furthermore, considering the security issues this would create i wouldn' t recommend taking such a path. On Wed, Jul 17, 2013 at 12:04 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is it possible to configure in the sip.conf for the Phone to be auto answer? Regards Bilal -- _ -- 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 -- Best Regards. Yasin SULUHAN Contact Information Mobile: +90 535 656 35 55 http://planetvoip.wordpress.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] 2GB Elastix memory limit
Hi... Since you use PAE kernel the server must be a 32bit machine i' m guessing this is a compiling issue... and i' m not sure how you can get past this issue... On Thu, Jun 28, 2012 at 11:58 AM, resea...@businesstz.com wrote: I have sevaral elastix installed but all of them show the physical memory is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE kernel but yet i cant see mem beyond 2GB. How can i configure the centos kernel to use more memory as the server is multipurpose Thanks Sam -- _ -- 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 -- Best Regards. Yasin SULUHAN Asterisk Telephony Infrastructure Consultant Contact Information Mobile: +90 535 656 35 55 Blog: http://planetvoip.wordpress.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] 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...
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
