Vodia support

Check this article on creating a extension Csv http://www.pbxnsipsupport.com/index.php?_m...kbarticleid=675

Bill you can set "Keep CDR Duration" for a 5 sec for example, This will help keep things under control. This is setting will enable the system to delete the cdr after 5 seconds. Article on Keep CDR Duration (smhd): The Keep CDR Duration (smhd) allows you to choose the length of time of the PBX keeps Call Data Record in the database. The Duration is in seconds, minutes, hours, days = (smhd). It is Important to note that if you set this "Keep CDR Duration (smhd)" to a long period, for example 100d (100 days), you will be caching and displaying 100 days of call data records. This is CPU intensive and can affect your system performance and use up disk space. Note that you have set only one of the 4 options (seconds, minutes, hours or days). You can not mix them. Examples of valid data: 7d or 6h or 45m or 30s etc. Examples of invalid data: 7d6h or 8h30m etc.

You can always use the (Direct Destinations) in the AA options and redirect them to a operator for example.

Can you provide us with the initial invite to the system? I justed tested on our system and the AA Does not hang up even If I press the wrong extension or random dtmf.

Can you post the sip invite to the hunt group?

When I usually upgrade to the latest I 1st start by upgrading gradually from version to version. try this chain of upgrades. http://www.pbxnsip.com/cs410/update-3.4.0.3201.tgz Not sure what version your running? Then try http://www.pbxnsip.com/cs410/update-4.0.1.3499.tgz To check the msp upgrade log into the ssh and try command cat/ etc/sipfxo-release This cmd should show you the the msp version.

This an article explaining how to update your msp and below that is the actual file. http://www.pbxnsipsupport.com/index.php?_m...kbarticleid=582 http://pbxnsip.com/cs410/update-msp.tgz

Not sure if this helps I haven't tried it myself. http://knowledgebase.polycom.com/kb/search...%200%2013302966 Copy paste to your URL.

Lofile should look like this. NOTIFY sip:42@192.168.0.34:2050;line=wci1bz2y SIP/2.0 Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK-d6a8dd9eed186e03416b38e516dc8190;rport From: <sip:42@pbx.company.com;user=phone>;tag=05f96e8ce6 To: <sip:42@pbx.company.com>;tag=al0f66n7nn Call-ID: 3c2693c7c50a-1s5kcf653dy3 CSeq: 16431 NOTIFY Max-Forwards: 70 Contact: <sip:192.168.0.33:5060;transport=udp> Event: message-summary Subscription-State: active;expires=355 Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:842@pbx.company.com Voice-Message: 3/0 (0/0) [7] 2010/06/10 20:09:06: SIP Tx udp:192.168.0.34:2050: NOTIFY sip:42@192.168.0.34:2050;line=wci1bz2y SIP/2.0 Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK-39949aef1b7405665f896b6d148673bc;rport From: <sip:42@pbx.company.com;user=phone>;tag=05f96e8ce6 To: <sip:42@pbx.company.com>;tag=al0f66n7nn Call-ID: 3c2693c7c50a-1s5kcf653dy3 CSeq: 16432 NOTIFY Max-Forwards: 70 Contact: <sip:192.168.0.33:5060;transport=udp> Event: message-summary Subscription-State: active;expires=355 Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:842@pbx.company.com Voice-Message: 3/0 (0/0)

The cisco phone has to subscribe to MWI. Can you call the cisco phone and leave a message and post the logfile on here.

If the trunk has the rate tab on top then you might have it. If not send your mac address to sales@pbxnsip.com along with this forum link.

Check out this article on Yealink provisioning. http://www.pbxnsipsupport.com/index.php?_m...kbarticleid=662

Which version are you using? What happens if you go directly to the option 5? do you get the same Greeting?

Check out http://www.pbxnsip.com/docs/pbxnsip_adminguide_v4.pdf page 277 chapter 16 Pre pay feature. The Prepay feature does require an additional license. Once acquired you will see a rates table in the trunk level.

check your provider cost 1st then configure your (dial plan preference) based on the least cost route.

Hi Matt we are currently cleaning this up.

I have tested this in the lab has describe above, I also tried different scenario. The end result is that all the extension in the group ring according to what is set in the duration, some range from 20-30 having from 3 to 5 extension and escaping to the AA for example. hey pbxnsip can you provide him with the latest beta?

LOL!!

These Kirk model is pretty Rugged nothing fancy about them, cool thing about is the you register to diffrent pbx's on one phone Robust, well-designed and competitively priced handset The KIRK 4020 handset is a robust, well designed and price competitive handset. It meets demands for free mobility and is built for long-term dependability in harsh environments. http://www.polycom.com/products/voice/wire...20_handset.html

Its very diffcult to recreate such occurance when a trunk disappears on your pbx. Its difficult to pin point in such rare cases but if it happends again please email the xml file to support@pbxnsip.com

Hey this is great stuff!!! Do you have anything for windows?

Not sure that will work, if you add an asterisk and then a extension. it will act more like a feature code.

I know when you press * during an announcement it will take to "please press your extension number"

Hi Matt We use sangoma A200 Cards here at Head Quaters and they work just fine, although the Netborders software contains the Wanpipe application as well. Pbxnsip does support FXS's only FXO's http://wiki.sangoma.com/wanpipe-windows-nbe-pbxnsip

Not Clear on this you want to use * as an Extension?