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I have been testing the part with ACD - I still cuts the call after apx. 90 mins. calling....

 

Minutes or seconds?

 

Try 4.0.0.3212 (send an email to support to get the latest snapshot).

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Sorry, 90 secs. :-)

 

Good day,

 

Juste a quick note, 4.0.0.3212 Win32 on our site is working fine but we cannot see call logs reports and/or Active calls.

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Juste a quick note, 4.0.0.3212 Win32 on our site is working fine but we cannot see call logs reports and/or Active calls.

 

Same here <_<

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does v4 support video now? I though that was one of the v4 items.

 

That part has not been touched yet, but 4.0 is not finished yet ;)

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Good day,

 

Juste a quick note, 4.0.0.3212 Win32 on our site is working fine but we cannot see call logs reports and/or Active calls.

 

 

Do we have anything new on that?

Does V4.0.0.xxxx Win32 newer then 4.0.0.3212 Win32 can be available for testing?

 

many thanks for the great job.

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Hi there, can you tell me if in the latest V4 beta this problem has

been addressed:

Forum Post

If so; please let us know where we can download latest beta release

 

We are currently testing the latest beta. There were a lot of "under the hood" changes and we simply need to test them. It does not make sense to give out a beta that will just have simple bugs.

 

3.4 is a great version; we don't see the immedate pressure to release/hand out something instable...

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We are currently testing the latest beta. There were a lot of "under the hood" changes and we simply need to test them. It does not make sense to give out a beta that will just have simple bugs.

 

3.4 is a great version; we don't see the immedate pressure to release/hand out something instable...

 

I understand completely but I see a downloadable version of 4.0 beta placed in the forum on June 11th. So I'm curious if in the beta this functionality is in.

Can you give me an estimate when a newer beta will be released.

The reason for this is that we are having a lot of troubles as we are only using softphones (OCS communicator or attendant) and after 20sec calls are dropped.

So especially with transferring calls we loose telephone calls.

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Where can I Downlaod the latest version ?

 

I understood that it also should be possible after forking to divert a call back to the extenios on the PBX. How does this work and is this already in this beta version ? (similiar to twinning on avaya ip office)

 

Thanks

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Where can I Downlaod the latest version ?

 

I understood that it also should be possible after forking to divert a call back to the extenios on the PBX. How does this work and is this already in this beta version ? (similiar to twinning on avaya ip office)

 

Thanks

 

Could you please reply on this

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Does this release resovle issues with VM hanging in mailbox and not sending email?

 

IE. We use 10 exts as some 100+ VM per day exts and about 2-10% of the items in 3.4.3201 hang in the mailbox and are not (we can tell easily as the VM is set to delete .. howerver email failures are not logged?)

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Does this release resovle issues with VM hanging in mailbox and not sending email?

 

IE. We use 10 exts as some 100+ VM per day exts and about 2-10% of the items in 3.4.3201 hang in the mailbox and are not (we can tell easily as the VM is set to delete .. howerver email failures are not logged?)

Not very clear on about the question. But there are couple of bugs have been fixed related to the voicemail/mailbox.

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When will PBXnSIP v4 be relseased - hope it will fix the problems I'm having with this great iPBX

 

I have a few Aastra 6757i CT Phones that need to be setup as a key system. The phones have 4 line buttons and that is all we require four our small business. The goal is to allow someone put a call on hold on one extension and allow it to be picked up on another extension. PBXnSIP shows that they only support SNOM phones with this type of setup - however we have used the aastra phones with another service and like the setup of the aastra phones over the snom phones. That being set we need these phones setup with shared call appearances as some call it and I would like to do it with PBXnSIP because I have researched many PBX systems and PBXnSIP is the best overall for features with main one problem - key emulation. If anyone can assist in or has documented success in setting up my Aastra 6757i phones properly for key system emulation please contact me. I would be willing to compensate for this help. The other problem we have is with DTMF - it is hit or miss as to when it works making it impossible to navagate phone menu's when calling Credit Card Co, even My sip provider SOTEL - I have had the problems with aastra 6757i, snom 820 & a pap2 adapter. We are using PBXnSIP version 3 hosted at I/O data center by IT Logistics - Please Advise!

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The latest Windows snapshot can be found here:

 

http://www.pbxnsip.com/protect/pbxctrl-4.0.1.3408.exe

 

Note: Please do not use this version in a production environment. We changed a couple of things that have not been tested well. For example, this build uses the SSE instruction set and uses otherwise optimized code. The Windows builds were slower than Linux build and we suspect this was because of the compiler options what we were using. So it would be great if you can help us giving feedback on the stability of this version, bt be careful with production systems.

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Hey, want to include a feature list we should test?

 

It would be interesting to see if there is any performance difference. Feature wise, cell phone forking for the groups, trunk CDR (including quality metrics), automatic blacklisting and maybe IPv6 interop (phase 2) should be the big topics.

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automatic blacklisting is going to cause a lot of service requests...

 

I agree... But I do believe in this feature and that it contributes to the system sanity. Maybe we should put bigger warning signs on the admin landing page (emails are being sent out already, but that is obviously not enough). We already made a timeout on the blacklisting so that eventually after one hour the system allows the next attempt (by default).

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I'm looking at cdr text file and see no difference?

 

You must send the CDR out as email. Right now you do this by using the scheme "mailto" in the global setting "soap_cdr_adr" (well, right now; we obviously need to change either the name or add a new setting); for example "mailto:matt@test.com".

 

Then the PBX will include an attachment with the MIME type "application/vq-rtcpxr". This will look like this (see http://tools.ietf.org/html/draft-ietf-sipp...rtcp-summary-06):

 

VQSessionReport: CallTerm
LocalMetrics:
TimeStamps:START=2010-01-07T13:28:58Z STOP=2010-01-07T13:29:51Z
CallID:0078-0B66-2DD33E69-0@D141DFC50C7AA2248
FromID:"00972781202229" <sip:00972781202229@domain.com;user=phone>;tag=007A-11C4-2CCCD3F2
ToID:<sip:423@domain.com>;tag=5c6f4cceba
LocalAddr:IP=192.168.0.233 PORT=55882 SSRC=0xcf528dde
RemoteAddr:IP=192.168.0.248 PORT=17264 SSRC=0x1e9f6401
x-UserAgent:pbxnsip-PBX/4.0.1.3409
x-SIPterm:SDC=OK SDR=AN
PacketLoss:NLR=0.0 JDR=0.0
BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=52123 GMIN=16
Delay:RTD=16 ESD=0

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