Re: [asterisk-users] Call Queue advise

to do so. my current thought is a Macro the queue runs at the answer of a call. Currently agents do not log in and out. That is slatted for roll out after a few more hires to that team. Thanks, -Bryan Anderson On Wed, Dec 11, 2013 at 10:22 AM, Chad Wallace cwall...@lodgingcompany.comwrote

[asterisk-users] Call Queue advise

current thoughts are probably a macro that places a penalty on the user tell the next call is answered. Any advice for this would be greatly appreciated. Thanks, -Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Call Queue advise

yes but I believe that least recent would ring one agent at a time? If my understanding is incorrect please correct it. We are wanting to keep with multiple phones ring to ensure coverage. Thanks, Bryan -Bryan Anderson On Mon, Dec 9, 2013 at 4:11 PM, Chad Wallace cwall

[asterisk-users] Homer SipCapture

Is anyone using Homer from sipcapture.org or anything like it for capture sip traffic for debuging? If so what are your experiences. Thanks -Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Users.conf vs Sip.conf

since it provides more flexibility. If I could get some help/clarification/advice as to the ideal setup between the two files to: 1) create users and, 2) provision phones I would greatly appreciate it. Thanks, -Bryan Anderson

Re: [asterisk-users] Polycom SPIP config

Ok, thanks for the info. -Bryan Anderson On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Thu, 7 Mar 2013 17:12:47 -0800 Bryan Anderson shadow...@gmail.com wrote: Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones

[asterisk-users] Polycom SPIP config

a way to stop the scrolling and just display the time? If so could you provide me the configuration parameter? thanks, Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

[asterisk-users] DST offset

Hello, I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting on the polycom phone provisioning template and I see that it has a variable for ${DSTOFFSET} in the template I tried adding dstoffset = -28800 to users.conf and sip.conf under both general and individual users but can't get the

Re: [asterisk-users] DST offset

to users.conf and sip.conf under both general and individual users but can't get the setting to set. Where/How do I set the variable? Thanks, Bryan On Mon, Feb 25, 2013 at 10:44 AM, Bryan Anderson shadow...@gmail.comwrote: Hello, I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting

Re: [asterisk-users] DST offset

ok, so after digging through google and the wiki and not finding anything I went through the code for res_phoneprov and found my answer. in users.conf set timezone= to the full name of the time zone. -Bryan Anderson On Mon, Feb 25, 2013 at 1:35 PM, Bryan Anderson shadow...@gmail.com wrote