similar to: advice on OrderlyStats (or other cc software)

Hi Everyone, There are a few things I like in OrderlyStats, specially some graph presentations and the fact that if agent puts someone on HOLD or PAUSE it shows fine. 1 -But I see a lot of similarities in pricing, descriptions, wording on both sites. Were these same projects forked out? or is it still both owned by the same company? 2- What are the main differences between the two? - so that we

Hello Voipcrazy, It's funny you should mention that - we've just released (as in today) a FREE version of our OrderlyStats service for call centre and queue monitoring and management. OrderlyStats features realtime (synchronous/message-based) display of all queue, agent and caller events so you can see what's happening in your call centre as it happens. The control panel feature

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hi Gabriel, Yes this information is shown in real-time and also in historical reports with the OrderlyStats system. OrderlyStats is now available as a Server Edition you can download and install yourself, as well as the FREE managed service. You can get it at http://www.orderlyq.com/statistics.html Hope this helps, Matt. Gabriel Ortiz wrote: Hi all, Is there a way to get the time

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Is there a way to have markdown automatically convert obvious (http, mailto) URL's to links? i.e: http://example.com -> <a href="http://example.com>http://example.com</a> Thanks, -- http://www.critikart.net

Hi, If several phones register to the same sip.conf section what will happen with a "Dial SIP/shared" in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy!

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, In our backup script we sometimes would like to override the common (i.e: static) excludes filter list. For example we exclude "- *.ext" for all backups but would like to include "+ *.ext" only for 'local' backups. Are such entries supposed to cancel each other? How can one override an earlier exclude in a filter list? Thanks,

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

If like me you run * on a VPN (or multihomed) gateway and want to serve remote SIP clients, make sure you have bindaddr = 192.168.0.1 ; or whatever is your box's private IP otherwise * might bind to its public IP and send it as return address in the SIP call setup, which will (should) be rejected by your firewall. To * experts: might this setting interfer with NATed SIP clients? -- I

Hi, After upgrading to 1.5.2 I no longer can directly access to my voicemail by pressing the "Message" button, I have to go through the "urgent,new,old" report first. The oneTouchVoicemail parameter is set to 1 but not taken into account apparently. Anyone noticed that problem?

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

Hi, In the samba debian package there used to be a very convenient utility to scan networks for open samba shares and weak passwords. Its name AFAIR is nat. Where can one find this tool? Is it still maintained? Is there a better alternative? Thanks, -- THERAMENE: Il veut les rappeler, et sa voix les effraie ; Ils courent. Tout son corps n'est bient?t qu'une plaie.

Hi, It seems rsync no longer resumes partial transfers after a SIGINT (CTRL-C). I tried the following: % rsync -avzP ~/video/Gone_In_60_Seconds_-_DivX.avi 192.168.0.3:/backup/DivX building file list ... 1 files to consider Gone_In_60_Seconds_-_DivX.avi 262144 0% 10.34kB/s 19:40:22 [CTRL-C] [testing remote size] % rsync -avzP 192.168.0.3:/backup/DivX/Gone_In_60_Seconds_-_DivX.avi