Vodia support
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Posts posted by Vodia support
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it doesn't hang up it just says "this ext dos not exists" and blank (you can enter a ext number if you know it the system doesn't restart the IVR)
You can always use the (Direct Destinations) in the AA options and redirect them to a operator for example.
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Hi
if a caller selects a wrong option the system we say "this ext number does not not exist" and the message does not start again
can you please fix this we have customers(online stores) complaining that there customers get lost and don't call again
please advise if there is a work around for now
Thanks
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.
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Hi
I received a sample a Panasonic sip Cordless Phone KX-TGP500B04
when call a Auto Attendant and then dial the ext number the phone will ring . well if i dial a hunt group and that ext is a part of the hunt group it doesn't ring
if i enter the ext number by Final Stage after is rings 30 sec and it hits Final Stage it starts ringing
I tried it with versions 3.5.0.3301 (Linux) 4.0.0.3344 (Linux) 4.1.0.4012 (Win32) 4.1.0.4026 (Win32)
did anyone ever have such a problem with any model phone? if yes can you please post what settings might help me
is their any different sip invite if coming through a hunt group?
Please advise
Thanks
Can you post the sip invite to the hunt group?
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What might we be doing wrong preventing us from upgrading the MSP and FIRMWARE on several CS410's?
Can you manually copy the firmware into a folder?
previously mentioned in http://forum.pbxnsip.com/index.php?showtopic=3664
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.
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If necessary, can an update package be manually copied onto the CS410, and if so what folder would you copy the update file into? Same folder for the MSP-UPDATE too?
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
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i would like to load a ring tone as a door bell into the PBX
i have polycom phones and a cyber data door bell and i would like to hear a difrint ring tone such as door bell ring tone when getting a call from the cyber data
can any one explain how to go about it
Not sure if this helps I haven't tried it myself.
http://knowledgebase.polycom.com/kb/search...%200%2013302966
Copy paste to your URL.
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Is there a trick to get MWI working? cisco blah
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)
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Is there a trick to get MWI working? cisco blah
The cisco phone has to subscribe to MWI. Can you call the cisco phone and leave a message and post the logfile on here.
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Hamlet,
Thanks for responding, any idea does the evaluation license come with prepay feature license? would like to try out before acquired.
regards,
Kelvin
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.
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Hi,
is there any manual how to setup provisioning for Yealink VOIP Phones ? they are listed as suppoerted phones but i cannot find any manual how to use it
AND
Is it possible to show the domain phonebook on the phone itself ?
Check out this article on Yealink provisioning.
http://www.pbxnsipsupport.com/index.php?_m...kbarticleid=662
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Which version are you using? What happens if you go directly to the option 5? do you get the same Greeting?
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Hi Support,
The version 4 prepay feature sound good. May i know how to enable this feature? does it require additional license? Regard the credit control does it apply to domain level instead of account level. It mean entire domain sharing same credit amount. Thanks
Regards,
Kelvin
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.
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my provider has 6 different routes i can choose from in my trunk. does the pbx has some kind of list cost routing so it will always choose the cheapest route?
thanx
check your provider cost 1st then configure your (dial plan preference) based on the least cost route.
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when i login as console user for ext 101 on IE 8 WAC shows conference calls great. When I do on Safari (mac) it doesn't even show the conference extension under the header. Also, the "Call Queue" header gets wrapped.
Matt
Hi Matt we are currently cleaning this up.
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i am running version 4.0.0.3344 and i have setup when a call comes in it reach first a hunt group that rings 5 Ext and have setup stage one for 10 second and Behavior final stage to go to an IVR and the PBX will only send it to IVR after 30 second no matter what i enter in stage 1
please advice, is this a bug or am i doing something wrong
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?
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I guess it depends on the hardware, it was the only one I was able to get working properly on the non toch screen windows mobile devices, such as my moto q.
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
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it has happened to me as well, i have a ticket open with pbxnsipsupport for several months and they were never able to figure it out, i also told them that i believe it happened when a System Admin was logged in, along with a Domain Admin
try checking other domains, and i wouldn't be surprised if the trunk ended up there "some how"
hopefully there is a fix for it
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
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Hey this is great stuff!!!
Do you have anything for windows?
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I dont understand your answer
this customer wants to use * as a Input and have the destination be an account
Not sure that will work, if you add an asterisk and then a extension. it will act more like a feature code. -
Not Clear on this you want to use * as an Extension?
I know when you press * during an announcement it will take to "please press your extension number"
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Has anyone used one of these?
(I know about http://wiki.sangoma.com/wanpipe-windows-nbe-pbxnsip.)
Are the FXS's useable to pbxnsip?
tx
matt
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
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Can i use * as a user input and have the destination be another Account? currently i have it set up but when i press * nothing happens, just silence
Not Clear on this you want to use * as an Extension?
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what information am i using to put into asterisk? they are trying to register to a sip gateway etc
anyone has this set up can help?
HI 911 did you configure the trunks yet?
LOL!!
Not sure that will work, if you add an asterisk and then a extension. it will act more like a feature code.
Odd Behavior CS-410
in Embedded
Posted
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.