natedev

While small companies are still mostly 32-bit, enterprises are predominantly running 64-bit operating systems. In fact, Windows Server 2008 R2 doesn't have a 32-bit version. While I recognize that there's not going to be a major advantage for pbxnsip to be 64-bit native, I think that it's important to move in that direction and to do lots of testing on 64-bit Windows Servers. My company is relatively small and out of our 60 servers, not one is 32-bit. -Nate

Editing pbx.xml and removing the "s" in "https" for the help URL fixed my issue but hopefully pbxnsip will address this for the next beta.

While I agree that the files should be secured by the OS file system, I think pbxnsip needs to store password hashes instead of the passwords. The problem is that most users like to reuse passwords so they don't have to remember a large list of logins. As a result, a system administrator may be able to see a password that could gain them access to something like an accounting system (where they're not the administrator). It's just not a best practice from an IT perspective.

I also noticed that each extension's passwords and PINs are unencrypted in the XML files in the /extensions folder.

Other issues: In the pbx.xml, the <email_pass> is unencrypted. The polycom_sip.xml and polycom_phone.xml files that are generated by pbxnsip are not up to date. The latest versions can be downloaded from http://downloads.polycom.com/voice/voip/sp...release_sig.zip http://downloads.polycom.com/voice/voip/sp...e_sig_split.zip and are named phone1.cfg and sip.cfg respectively. Would like to have the phone and sip template files that are generated by pbxnsip in a folder called /templates rather than hard-coded into the program. If you guys change parameter names from version to version, it breaks any modified files that may have been placed in /html. When I go to the online help, it appears as though some resources are SSL and others not so I get a security warning from IE saying "Do you want to view only the webpage content that was delivered securely?". Using all relative URLs will solve that issue. I see that I can fix it in pbx.xml by changing "https" to "http" in <app_help>. Online help - I would recommend expanding all acronyms in the help area. There's an assumption that users know what PSTN, ACD, CMC, CID, IVR, ANI, CDR, etc. mean. While many advanced users will know some, it's unlikely that they know all of them.

NTP Port really shouldn't be associated with the TFTP ports section. Also, I don't see how to explicitly specify an NTP server other than editing pbx.xml's <ntp_host>. I'd like to be able to specify more than one NTP server, too.

I'd like to see help built into each version rather than put on pbxnsip's servers. The problem is that the pbxnsip hosted help has to be version independent and the screenshots are always off. Instead, I recommend using a Web 2.0 effect that looks like a dialog that would appear within the page with the help for fields on that page. It could be section by section since each has a ? button. This would be more elegant than opening a new window.

Has anyone ever had the "Remember login information" checkbox ever work for them in any version of pbxnsip? I've used IE, FireFox, Chrome, and Safari and never seen it work even though cookies are enabled on all of them. Also, I noticed in the 4.0 beta that I sometimes have to login twice to get in (but no error about bad password).

After some more testing, we found that all calls from our SoundPoint IP 670s (internal) to an extension would connect properly to the cell phones. Only calls coming in from the Broadvox trunk fail to connect to the cells.

We just did some testing. It's intermittent. I had one of my staff call our main number (Broadvox trunk) from their cellphone. That one forked to my cell properly. Then, the next 3 calls didn't fork properly.

Thanks for replying. How did you diagnose and resolve it? Did you change the Codec Preference on Admin -> Settings -> Ports or specify an Override Codec Preference in the Domain -> Trunks -> Edit? What type of SIP phone are you using?

I have pbxnsip 3.3.1.377 (Win32). On Domain -> Accounts -> Redirection, I have the following: Cell phone number: <my cellular number> Confirmation: No confirmation required Include the cell phone in calls to extension: Immediately Specify time when system calls the cell phone: No specific time excluded When I get incoming calls, both my Polycom SoundPoint IP 670 and my cellphone ring. The cellphone even shows the correct caller ID. However, if I answer the call on my cellphone, I get nothing. The call appears to be connected but there's no audio in either direction. I can make and receive calls on my SoundPoint to the outside world so there doesn't appear to be a problem with my Broadvox trunk. Any suggestions as to what might be causing this? I'm answering the cell on the first ring. Thanks in advance. PS: this has been a problem for me going back a while.

Thanks! Here's a modified version for Polycom SIP 3.1.2RevB: polycom_phone.xml I used BeyondCompare to edit it side by side with your template.

No - it was the polycom_sip.xml rather than the polycom_phone.xml.

Would it be possible to get the polycom_phone.xml template that is built into pbxnsip 3.3.0.3165? I noticed that it's based on Polycom SIP 3.1.1 and I wanted to create a version based on the phone1.cfg that comes with Polycom SIP 3.1.2RevB. Thanks in advance.

FYI: This was entered into OTRS as Ticket 2009031910000041. It is still open.

Since upgrading from 3.2.0.3144 to 3.3.0.3165, the Address Book (accessed from the physical phone by pressing the Directories button then "1. Contact Directory...") stopped working on our SoundPoint IP 670 phones. These are running with Polycom SIP 3.1.2RevB and BootROM 4.1.2RevB (the latest of both). I've tried the following: - Emptied my /html folder (previously, I did have a polycom_sip.xml which I created starting with the template that is used for 3.3.0.3165 so there weren't many deviations in the first place - none related to the address book) - Restarted the pbxnsip service - Reset my phone (Reset to Default -> Reset Local Config... and Reset Device Settings... then set it up to provision via username and password) I was having this issue with some of the pre-release builds of 3.3 as well and it was suggested that .3165 should solve the problem (but it didn't). Any thoughts or suggestions?

Here's my revised template based on Polycom SIP 3.1.2RevB: polycom_sip.xml

Thanks a lot for posting this! It's very helpful. If you'd like, I can parameterize Polycom's one for SIP 3.1.2RevB and post it here. It would be great to have that be part of the new builds. I do see that several parameters were renamed such as {sip-port layer} to {sip-port polycom_layer}. That explains why my polycom_sip.xml wasn't working once I upgraded to 3.3 from 3.2.

Do you have an updated version of the template for 3.3.0.3165? Thanks!

For a small percentage of our outbound calls each day (about 3 or 4 times per day), we'll hear ringing and then suddenly a fast-busy sound and the call drops. Our SIP trunk provider is Broadvox and they did a capture of the problem when it occurred. What they said is that sometimes pbxnsip 3.2.0.3144 (win32) isn't sending a response to their 200 code to indicate acknowledgement of the connection. As a result, Broadvox thinks the caller on the pbxnsip side hung up so they disconnect. Is anyone else running into this? If so, what version are you running (and what OS)?

I went to that page in the Wiki. I see the link to http://doc.otrs.org but it just seems like a manual for OTRS. Now I know how to install OTRS on our server. I'm not sure how to connect to your installation of OTRS so I can log these issues on your system. Do you have a link you could send me? Do I need a login? Thanks in advance.

No - nothing like that. When I call in from the outside through the trunk, I am connected to the auto-attendant correctly. We have options like press 1 for sales, 2 for tech support, 3 for HR, 4 for accounting, 5 for dial-by-name, 0 for the operator or enter an extension directly. The inputs are 1#, 2#, 3#, 4#, 0# and 5# for dial-by-name. I'm entering 100 very quickly but it connects me to 201. If I enter 201, it connects me to 100. Normally, when I enter an extension, it announces the name. There's some clipped audio when I enter the extension on the 3.3 builds. So it could be an issue with the auto-attendant. The only thing is that that when I call the auto-attendant internally, it routes calls correctly. It's only when the call comes from a trunk that it goes awry. As soon as I revert to 3.2.0.3144, it all works again.

Could I get some feedback on the items I identified in that build? Do you maintain lists of defects that are resolved in each new build?

Just tried 3.3.0.3156 (win32). The following appear to still be issues: - I just tried calling in to x100 from my cellphone and it connected me to x201. The log made it look like I was calling x201 even though that's not what I dialed. Calling x201 rang x100 (my phone). But if I call from my extension to x201, it works fine and vice versa. There's nothing in the logs that suggests an error. They make it look like I was dialing the extension that I was connected to. This is a really serious problem. - The address book isn't working on my Polycom SoundPoint IP 670. - If I place my own parameterized polycom_sip.xml in the /html folder, the generated polycom_sip.xml is missing lots of values. If I remove that file, it generates a polycom_sip.xml based on Polycom's SIP 3.1.1 instead of 3.2 (which is what my file is based on). Did the parameter names get changed in pbxnsip 3.3? Seems like that's what happened. Is there an updated list of the parameters or can I get your source file for your parameterized polycom_sip.xml? I think you might want to not have these hard-coded in the future. Why not just drop them into /html ? - Settings -> PnP: the field labels are still missing. I would have thought these are based on the files in /pnp_parms. Fixed - I am able to modify the MAC address field on the Account -> Registration page now.