Vodia PBX Posted December 17, 2008 Report Share Posted December 17, 2008 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! Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted December 17, 2008 Report Share Posted December 17, 2008 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. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 18, 2008 Author Report Share Posted December 18, 2008 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. Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted December 19, 2008 Report Share Posted December 19, 2008 Just to validate the revision for debian is 3109 not 3110 .. is that correct? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 19, 2008 Author Report Share Posted December 19, 2008 Just to validate the revision for debian is 3109 not 3110 .. is that correct? Yes, something went wrong. Until we find the 3110 build, there is a http://pbxnsip.com/download/pbxctrl-debian4.0-3.1.1.311 build. Quote Link to comment Share on other sites More sharing options...
pbx support Posted December 19, 2008 Report Share Posted December 19, 2008 Yes, something went wrong. Until we find the 3110 build, there is a http://pbxnsip.com/download/pbxctrl-debian4.0-3.1.1.311 build. It is http://pbxnsip.com/download/pbxctrl-debian4.0-3.1.1.3111 Quote Link to comment Share on other sites More sharing options...
andy Posted December 20, 2008 Report Share Posted December 20, 2008 Probably found a bug with the mailbox. Please check my post under Mailbox for details. Moving back to an older built solved the problem immediately. Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted December 24, 2008 Report Share Posted December 24, 2008 . Quote Link to comment Share on other sites More sharing options...
natedev Posted January 1, 2009 Report Share Posted January 1, 2009 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 1, 2009 Author Report Share Posted January 1, 2009 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? Quote Link to comment Share on other sites More sharing options...
natedev Posted January 6, 2009 Report Share Posted January 6, 2009 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. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 6, 2009 Author Report Share Posted January 6, 2009 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? Quote Link to comment Share on other sites More sharing options...
natedev Posted January 6, 2009 Report Share Posted January 6, 2009 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. Quote Link to comment Share on other sites More sharing options...
natedev Posted January 7, 2009 Report Share Posted January 7, 2009 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. Quote Link to comment Share on other sites More sharing options...
natedev Posted January 7, 2009 Report Share Posted January 7, 2009 Just downloaded 3.1.2.3120. I see the general release notes for 3.1. Is there anything information about the changes made in this latest build? Thanks, Nate Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted January 7, 2009 Report Share Posted January 7, 2009 Just downloaded 3.1.2.3120. I see the general release notes for 3.1. Is there anything information about the changes made in this latest build? Thanks, Nate Scroll down, you will see the 3.2 release updates. Quote Link to comment Share on other sites More sharing options...
kcnd Posted January 7, 2009 Report Share Posted January 7, 2009 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! Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted January 9, 2009 Report Share Posted January 9, 2009 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!! Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 9, 2009 Author Report Share Posted January 9, 2009 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. Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted January 9, 2009 Report Share Posted January 9, 2009 So, you are saying the payload changed to use 101 now? Also, it would be wise to document the default payload id as most default on 96. Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted January 14, 2009 Report Share Posted January 14, 2009 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 Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted January 28, 2009 Report Share Posted January 28, 2009 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. Quote Link to comment Share on other sites More sharing options...
pbx support Posted January 28, 2009 Report Share Posted January 28, 2009 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) Quote Link to comment Share on other sites More sharing options...
pbx support Posted January 28, 2009 Report Share Posted January 28, 2009 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? Quote Link to comment Share on other sites More sharing options...
Carl Johnson Posted January 28, 2009 Report Share Posted January 28, 2009 Just remembered, We had some reports of this when using Polycom 2.2.2.x version. What phones are you using? That is the phone and SW version we are having trouble with. I will try 3.x SIP app and see if that helps .. any idea what the issue is? Quote Link to comment Share on other sites More sharing options...
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