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So we did build 3.1.1.3110 for all operating systems today and made them available from the pbxnsip software download page (http://pbxnsip.com/software). Although there are still a few open points, this should be the best version ever. Release notes can be found on the Wiki on http://wiki.pbxnsip.com/index.php/Release_Notes_3.1, and as you can see we did fix a lot of problems and also added some interesting new features.

 

The main point of 3.1.1 is that tel tel:-alias has been replaced with a telephone number. That makes it easier to represent that number in different environments, international or carrier-dependent and it also makes telephone number matches more safe. Also we added a new trunk identification mechanism possible that is very useful for multiple-domain environments. We made it possible to use the Cisco 7961 phone series with the pbxnsip in multiple domain environments, and even use features like park and pickup. And we found a ugly memory leak that could surface when Polycom phones are automatically provisioned with TCP transport layer.

 

Thanks for all those who were helping out testing the different builds in all kinds of environments!

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So we did build 3.1.1.3110 for all operating systems today and made them available from the pbxnsip software download page (http://pbxnsip.com/software). Although there are still a few open points, this should be the best version ever. Release notes can be found on the Wiki on http://wiki.pbxnsip.com/index.php/Release_Notes_3.1, and as you can see we did fix a lot of problems and also added some interesting new features.

 

The main point of 3.1.1 is that tel tel:-alias has been replaced with a telephone number. That makes it easier to represent that number in different environments, international or carrier-dependent and it also makes telephone number matches more safe. Also we added a new trunk identification mechanism possible that is very useful for multiple-domain environments. We made it possible to use the Cisco 7961 phone series with the pbxnsip in multiple domain environments, and even use features like park and pickup. And we found a ugly memory leak that could surface when Polycom phones are automatically provisioned with TCP transport layer.

 

Thanks for all those who were helping out testing the different builds in all kinds of environments!

I don't see the Windows update edition there. Using the full version as an update, does not work for me in the last few versions.

I did find it in the Protect folder, nice to see file size is smaller than previous version.

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I don't see the Windows update edition there. Using the full version as an update, does not work for me in the last few versions.

I did find it in the Protect folder, nice to see file size is smaller than previous version.

 

Ops, seems the link was wrong... Should be okay now.

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I applied the update (3.1.1.3110) on Windows 2008 Server x64. Whenever my Polycom SoundPoint IP 670's were powered on, they crashed just as they started the SIP application (going into an endless reboot cycle). We have BootROM 4.1.2.0037 and SIP 3.1.1.0137. I switched back to 3.0.1.3023 (Win32) and all is well again (other than the problem we reported with not being able to create multiple CO lines but that was happening in both versions).

 

-Nate

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I applied the update (3.1.1.3110) on Windows 2008 Server x64. Whenever my Polycom SoundPoint IP 670's were powered on, they crashed just as they started the SIP application (going into an endless reboot cycle). We have BootROM 4.1.2.0037 and SIP 3.1.1.0137. I switched back to 3.0.1.3023 (Win32) and all is well again (other than the problem we reported with not being able to create multiple CO lines but that was happening in both versions).

 

Whow. Can you send me the generated files in a PM for plug and play?

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Whow. Can you send me the generated files in a PM for plug and play?

 

Here's what's happening...

- I have "Username/password required" selected for TFTP -> Generate passwords.

- Under 3.0.1.3023, I can go to http://<my server's IP>/provisioning/polycom_phone_<my phone's MAC>.cfg and it'll prompt me to authenticate. I put in my extension number (only one domain) and password and it displays the XML configuration file no problem.

- Replace 3.0.1.3023 with 3.1.1.3118 and try the same thing and you won't get authenticated.

 

Because the Polycom's can't get their configuration when they're booting up, they try to reboot hoping this will cure the condition and the endless cycle begins.

 

If I change TFTP -> Generate passwords to anything other than "Username/password required", it works ok. Of course, I want username/password required.

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Here's what's happening...

- I have "Username/password required" selected for TFTP -> Generate passwords.

- Under 3.0.1.3023, I can go to http://<my server's IP>/provisioning/polycom_phone_<my phone's MAC>.cfg and it'll prompt me to authenticate. I put in my extension number (only one domain) and password and it displays the XML configuration file no problem.

- Replace 3.0.1.3023 with 3.1.1.3118 and try the same thing and you won't get authenticated.

 

Because the Polycom's can't get their configuration when they're booting up, they try to reboot hoping this will cure the condition and the endless cycle begins.

 

If I change TFTP -> Generate passwords to anything other than "Username/password required", it works ok. Of course, I want username/password required.

 

Well, you have to set up the the password on the Polycom phone (see http://wiki.pbxnsip.com/index.php/Polycom). Did you do that?

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Well, you have to set up the the password on the Polycom phone (see http://wiki.pbxnsip.com/index.php/Polycom). Did you do that?

 

Yep, did all of that. Here's what I've discovered. If Settings -> Ports -> TFTP -> "Generate passwords" is set to "Username/password" required", under 3.0.1.3023, you can login as your extension number and password. Under 3.1.1.3118, you have to specify the domain even if there is only one domain (eg 100@mydomain.com instead of 100). Would this qualify as a bug in that the functionality that was there was removed?

 

Also, the Polycom configuration files that pbxnsip generate (http://<pbx address>/provisioning/polycom_phone_<MAC>.cfg) set reg.<number>.auth.userId to the extension number and not extension@domain.

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Not sure if anyone else is running into this but when we add an entry to the Domain Address Book (Domain -> Settings -> Address-Book), all the existing entries disappear in the table above until the pbxnsip service is restarted. Also, if I change the result length from 10 to 20 or 50, it doesn't seem to remember that setting and always goes back to 10 which is too short, IMO, for a default.

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Issues discovered with 3.1.2.3120

Using this version, calls coming into an ACD/IVR account will not accept DTMF entry (using a Grandstream 4108 FXO for trunking). The last version that worked is 3.1.1.3101 which is what we have had to go back to. DTMF works fine for eveyhting else - just the ACD/IVR will not work.

 

Issues discovered with 3.1 - any version

Changes made to the handling of BLF notifaction to phones does not work with Grandstream phones (firmware 1.1.6.44 - latest release - or any earlier release). This always worked fine with version 2 software. BLF lamps will not work at all unless an * is placed in the extensions "Watch calls of the following extensions" and then the lamps will light with an initial status but never update again - i..e they light red or flashing red but never chnge status again. These same phones will work with version 2 software fine.

 

Option to turn off cell phone linkage

Is there an option in version 3 to turn off the cell phone linkage that detects a caller coming into the system via their cell phone and prompting different than the stadnard in-bound call paths?

 

Thanks for your help!

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3120 Release. Same issue as above DTMF that is transported in Out of band via RTP or Singaling does not work!!! Mediatrix PRI Gateway. Please fix and re-release!!

 

On the DTMF topic, we fixed a bug in the codec negotiation. Essentially the bug was that the PBX was offering the codec number of the foreign device, not it's own codec number. For example, when the PSTN gateway offers 96 for DTMF, the PBX would answer also with 96. However internally it uses 101. In the fix, it now answers with 101 (which is correct). The problem because obvious for outbound calls when the PBX has to generate a SDP offer. When the PBX offers 101 and the other side answers with 96, then the PBX has to send codec 96, and receive codec 101.

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VM seems to be a recurring issue with new releases, I found another issue with 3.1.2.3120 where a caller comes in and leaves a VM and the VM is sent to the recepient via Email but DOES NOT light the MWI but is in the phone mailbox?

 

1) The ext is setup to send MWI

2) The email is setup to email as attachment and keep as new message

3) Registration shows a message summary but the phone does not light?

5) Worked in 3.1.1

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  • 2 weeks later...

v3120. CS410 .. issues!

 

When using v3120 on a CS410 and a new call comes in (beeps in) while on a call the audio cuts out and the other end can hear me speak but I cannot hear them!! This was not true on the previous versions. Is this a known issue? Also, we are still having VM issues on v3120 with the VMs not recording properly and a key press of some sort needing to be used .. please address of PM me so we can work this issue out with whatever traces you need.

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v3120. CS410 .. issues!

 

When using v3120 on a CS410 and a new call comes in (beeps in) while on a call the audio cuts out and the other end can hear me speak but I cannot hear them!! This was not true on the previous versions. Is this a known issue? Also, we are still having VM issues on v3120 with the VMs not recording properly and a key press of some sort needing to be used .. please address of PM me so we can work this issue out with whatever traces you need.

 

Please PM to support@pbxnsip.com with SIP/RTP trace if you have. (This seems like a problem that has been solved already)

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v3120. CS410 .. issues!

 

When using v3120 on a CS410 and a new call comes in (beeps in) while on a call the audio cuts out and the other end can hear me speak but I cannot hear them!! This was not true on the previous versions. Is this a known issue? Also, we are still having VM issues on v3120 with the VMs not recording properly and a key press of some sort needing to be used .. please address of PM me so we can work this issue out with whatever traces you need.

 

Just remembered, We had some reports of this when using Polycom 2.2.2.x version. What phones are you using?

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