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Okay, 2.1.0.2108 is out

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See:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2108.exe

http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2108

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2108

http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2108

http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2108

 

It seems that the RTP pass-through problems are fixed. Also, this version now always sets the processor affinity to a value (1 is default now). Looks like this version solves a lot of problems.

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Now you need to put a file path there, e.g. img/logo_custom.gif located in the html/img/logo_custom.gif file relative to the working directory.

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See:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2108.exe

http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2108

http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2108

http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2108

http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2108

 

It seems that the RTP pass-through problems are fixed. Also, this version now always sets the processor affinity to a value (1 is default now). Looks like this version solves a lot of problems.

 

Paging is now working with relaease 2108 - I'm not sure what changed, but all is well now. I did not have to apply a new ringtones.xml file.

No issues came up from using 2108 under a normal business day on Friday, so it appears to be stable at this point. Have not had any more issues with DTMF and some of the echo issues we were experiencing have gone away (still have other echo issues related ot he phone devices, but pbxnsip appears to be clean).

 

ALso, some feedback on performance: we are using a server with two dual-core Xeon 3 Ghz processors, 4 GB RAM and 1GB NIC with Windows Server 2003 R2. One of the eight virutal processors is dedicated to pbxnsip via affinity control. Under typicall business office call loading, we see no more than 5% CPU utilization and 15% network utilization (several hundered calls per day with 25 key uesrs and 35 total users; 7 PSTN lines via a gateway with as many as 5 in use at any one time). Even under unicast paging of 25 phones, the CPU utilization doesn't spike higher than 15%. Seems release 2108 is quite efficient.

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Now you need to put a file path there, e.g. img/logo_custom.gif located in the html/img/logo_custom.gif file relative to the working directory.

 

 

oh you cant have a URL to an external site anymore?

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I am running 2.1.0.2108 and everything seems to be working great... One of the reasons I upgraded was to try out TAPI...

 

I have downloaded pbxtsp10.exe from the site and have it installed and properly setup in Control Panel... When I go to Dial in Outlook it just stays at Dialing and nothing happens...

 

The Click To Dial with the link works, but I'm trying to get the Outlook dialing working...

 

Is there anything else I should setup? Are there any other ports that should be open in the firewall? I left all the ports default...

 

thanks!

 

david

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Do yo usee SUBSCRIBE with the Event x-tapi showing up on the PBX? If not there must be something with the TSP setup. And you have to reboot your PC after installing the TSP and setting it up.

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This appears to be a Unified Messaging issue with build 2108 (unless it's also due to the lang_en.xml that we removed): before this build, missed call notifications and voicemail notifications had the correct caller/caller ID information attached to it. 'Missed call from Jack', or 'voicemail from 3015555555'. Now, all missed calls and voicemails appear to come from the user themselves, for example, all my voicemails say: 'Voicemail from Sean'.

 

We went straight from 2.0.3.1715 to build 2108 (and removing the lang_en.xml), so if this is a 2.1 bug I don't know how long it's existed.

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Nope, the file must be on the local file system now. Should not be a too hard requirement IMHO.

 

 

Still doesnt work. The XML is ok I can manually put in the URL and pbxnsip shows me the logo "http://pbx.com/img/logo.gif"

 

However pbxnsip will always show: http://pbx.com/img/main_logo.gif

 

i update it and update it and it never shows anything different.

 

db

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G722: Yes. Also known as codec 9. But you must list it in the codec_preference setting.

 

Image: I attached a screenshot of something that works.

customize.gif

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This appears to be a Unified Messaging issue with build 2108 (unless it's also due to the lang_en.xml that we removed): before this build, missed call notifications and voicemail notifications had the correct caller/caller ID information attached to it. 'Missed call from Jack', or 'voicemail from 3015555555'. Now, all missed calls and voicemails appear to come from the user themselves, for example, all my voicemails say: 'Voicemail from Sean'.

 

We went straight from 2.0.3.1715 to build 2108 (and removing the lang_en.xml), so if this is a 2.1 bug I don't know how long it's existed.

 

Is anyone else running UM alongside a PBXnSIP 2.1 RC version that can either confirm or deny that this is a problem with 2.1? That's when we started to see it.

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Just installed the .2111 and still have so far only two smaller issues:

 

- Music on hold is not working with the pre-configured moh.wav file. It's silent to the caller if a call is on hold or parked.

- The auto attendant does not announce "please enter the extension" after the customer recording ends.

 

Those worked before with 2.0.3.1715. The .2111 release was a fresh clean install from 2.1.0.2093 and upgraded to .2111

 

Detlef

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The missed call emails that I am getting are okay.

 

Are you using your own files in the html directory?

 

Thanks, yes there are a few -- we've been working alongside Model12 on some Polycom issues and so right now what I have in the html directory are four files: polycom_adrbook.xml, polycom_master.xml, polycom_phone.xml, and polycom_sip.xml.

 

The modifications that have been made by us to those files are all in the polycom_sip.xml file, and they are:

- a hardcoded dialplan: <digitmap dialplan.digitmap="[2-6]xxx|7xxxx|[2-9]11|91xxxxxxxxxx|9[2-9]xxxxxxxxx|*xx[2-6]xxx|xx.#" dialplan.digitmap.timeOut="3"/>

- a modded <CALL_WAITING tone.chord.callProg.6.onDur value

- switched the <volume voice.volume.persist.handset value to 1

 

Thanks for the help...

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Here we go with 2.1.0.2113: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2113.exe.

 

Changes: Some Email servers had a problem with the base64-encoded format, that was changed. Also the SOAP CC authentication (PIN) was not working. Credentials based on P-Preferred-Identity (inbound) were not working. The ACD IM messages could "stall". Linksys provisioning files updated.

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Just upgraded to 2114 and it fixed my email issues from 2108...

 

I have a question about codecs and my sip.cfg... Here is the section in my current sip.cfg file...

 

 

<codecs>

<preferences voice.codecPref.G711Mu="1" voice.codecPref.G711A="2" voice.codecPref.G729AB="3" voice.codecPref.IP_300.G711Mu="1" voice.codecPref.IP_300.G711A="2" voice.codecPref.IP_300.G729AB="3" voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/>

<profiles voice.audioProfile.G711Mu.payloadSize="20" voice.audioProfile.G711Mu.jitterBufferMin="40" voice.audioProfile.G711Mu.jitterBufferShrink="500" voice.audioProfile.G711Mu.jitterBufferMax="160" voice.audioProfile.G711A.payloadSize="20" voice.audioProfile.G711A.jitterBufferMin="40" voice.audioProfile.G711A.jitterBufferShrink="500" voice.audioProfile.G711A.jitterBufferMax="160" voice.audioProfile.G729AB.payloadSize="20" voice.audioProfile.G729AB.jitterBufferMin="40" voice.audioProfile.G729AB.jitterBufferShrink="500" voice.audioProfile.G729AB.jitterBufferMax="160" voice.audioProfile.Lin16.payloadSize="10" voice.audioProfile.Lin16.jitterBufferMin="20" voice.audioProfile.Lin16.jitterBufferShrink="500" voice.audioProfile.Lin16.jitterBufferMax="100" voice.audioProfile.Lin16.frequency="16000" voice.audioProfile.Lin16.payloadType="117"/>

</codecs>

 

 

I have a Polycom IP430... I want to use the best codec possible for sound, I have no preferences set on the Trunk.. Are the preferences in the above sip.cfg supposed to match with pbxnsip? If so what should they be set? What is a good default for trunk codec preferences?

 

Thanks..

 

david

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The PBX takes the freedom to resort the list of proposed codecs according to its internal preferences. As long as the user agent promotes all available codecs it is fine.

 

The 2114 build also has a global codec preference in the system admin/settings/ports section. Did you see that?

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