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 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, 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 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 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
Re: [asterisk-users] s/n ratio detection etc...
On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas 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
[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
