Bill H

OK, so messy is my middle name. This isn't the first thing I got into that was not easy. I have seen the files in the "Generated" folder and I understand Polycom configuration methods. My question is upstream from there: Where is the pnp.xml file???? I re-read the WIKI several times and now I am beginning to understand the PBXNSIP method to modify the file(s). But this is done in the pnp.xml and it (pnp.xml) can't be found. I even switched to 3.4.0.3201 (Win32) and it still isn't there. It isn't practical in this instance to use the TFTP server to create my own configuration files. Any ideas??? Anyone????

I am asking the same question, only for Polycom. http://forum.pbxnsip.com/index.php?showtopic=3193

I have read the WIKI/Professional Services webpage and I can't make any sense out of it. https://www.pbxnsipsupport.com/index.php?_m...kbarticleid=506 I am using Version 4.0.0.3204 (Win32) not sure exactly how I got it. Anyway, I don't see this From the WIKI: "When the PBX starts, it reads the file "pnp.xml". " I can't find a pnp.xml file. ----- Also from the WIKI: "For every file that should be processed during the plug and play configuration, you need to add an entry. The name tag in the file entry indicates which file should be read from the html directory. " OK this seems simple, but where is the "polycom_master.xml"??? This is from the pnp.xml (which I can't find): <file name="polycom_master.xml" encoding="xml"> <pattern>!^(0004f2[0-9a-f]{6})\.cfg!\1!</pattern> <vendor>Polycom, Inc.</vendor> </file> Is this the current method for Dynamically Changing PNP files or is there something newer? Or am I just reading it wrong?

I see your callflow scenario OK. Could you spell out the circumstances of the caller and the objective of the Auto Attendant and ACD Group.

Sorry too late now... We used a master pbx.tar file to set the PBX up. We just defaulted the PBX and re-loaded the config to get us back to square 1. The trunk was the only thing that was configured in the installation so it wasn't a big loss to start all over.

New installation with Version 3.4.0.3201 (Win32) 2 Callcentric SIP Registration (Trunks) were created and work OK. A SIP Gateway (Trunk) was created for an external service provider that did not require, or want, SIP Registration. During the process of setting up and testing, the SIP Gateway (Trunk) disappeared....... The XML file still has it there but it does not come up in PBXNSIP Admin screen for trunks. Any ideas as to how to get the trunk back?

We save the CDR as a CSV File. Is it possible to email the CSV file, as an attachment, as the "Send daily CDR report to:"

We installed a CS-410 for a new customerand we have been having some troubles that we have seen before. The basic trouble at this point is no disconnection of internal calls. (maybe outside too / both PSTN and SIP Trunk) If a call is made from one phone to a mailbox, on the system, it connects OK but when the caller hangs up the call does not release. This was verified by looking at the "Calls" page of the PBXNSIP Admin GUI while the LED on the phone stays on. I traced it to a missing BYE Message last week so I updated the firmware in all the Aastra phones. That seemed to clear it up. But then this week it returns so I updated the firmware in the CS-410 to the most recent from PBXNSIP website. Still no disconnect. The odd thing is that I have a remote extension off of his system and it works OK 100% of the time. We are running everything through one port on the CS-410 (Wan with SIP Replacement) and I am wondering if this may be linked to the trouble. What are your thoughts??

Has anyone used any additional Virus Protection on PBXNSIP running Windows? I mean software in addition to the basic Virus Protection that comes with Windows.

We have a customer with an unusual trouble. Twice now it has happened. The system runs fine for weeks then all of a sudden on two occasions, it goes bad. First, his modem/router lost all of its account settings, would not log on to the ISP and had to be re-programmed. Next, his PBXNSIP computer was sending an incorrect return IP Address in the SDP and we had to force it by placing the correct address in the SIP IP Replacement List: area (the network card was OK) Finally, the Audiocodes PSTN gateway, which has a static IP Address on the LAN, changed as if it were DHCP. OK, so we fixed all that, then it happens again...about a week or so later..The Internet goes out. (but for a different external reason this time) This time PBXNSIP's password is changed, the CPU is running at 50 to 100% consumption, the PSTN gateway has changed again. We reduced the network down to PBXNSIP, PSTN gateway and one phone plugged directly into the Netgear POE switch and the trouble (CPU fully loaded) persisted. Later that evening the customer calls and says everything is OK now. Has anyone seen this type of trouble before??

Are your PBXNSIP's behind a router?

I have it happening on Aastra phones. I looked and there was no NOTIFY message being sent from PBXNSIP to the Subscribing phones when the call was answered. Any fix yet???

Are the calls going out on PSTN lines????

We have a customer using a Cable Modem with a built-in FXS gateway. It looks like he is having the same trouble. CS_410 Version 3.3.2.3181 (Linux) Does this version include the Phantom Caller-ID fix?????

Is it possible to purchase mixed licenses? The mixed license would allow some extensions to have extended features and some to have basic features.

Same question, different gateway. Sangoma cards and the supporting software only allow a single port and IP Address to address up to 12 PSTN lines. Can anything be done using CO1 CO2 etc... in the PBXNSIP Trunks setup to allow selective use of a trunk??

Try: Select your trunk in the dial plan. Pattern: 7* Replace: *

I don't have any experience with the Patton. I did look at one (set up screens) a couple of times and it looks like one of the most confusing and difficult gateways to work with. Anyway, you would need to see if there is a way to partition the ports in the gateway into separate groups. Audiocodes and Grandstream will do this and I have been able to create 2 Trunk Groups or more in these gateways. Something like Trunk Group 1 (in the gateway) is 192.168.1.123 port xxxx on PBXNSIP Trunk 001 SIP Gateway and Trunk Group 2 is 192.168.1.123 port zzzz on PBXNSIP Trunk 002 SIP Gateway. The gateway Trunk Groups could be from 1 to 4 gateway ports. The IP Address will be the same since all Trunks are in the same gateway. Its the Ports that make the difference.... Then set your Auto Attendants account number in each of the respective trunks (001 002 003 etc...) Send call to extension: Bill H

I have thought about this also. I didn't see anything in the Trunk settings that could be labeled like "ABC Corp". My solution is to have the Trunk go to a Hunt Group (which can have a name like "ABC Corp") and the place any extension(s) in the Hunt Group and set From-Header: to Group Name. If someone has a cleaner method, please let us know..... Bill H

Thanks, it works OK now.

I want to set the PRIORity of PBXNSIP but, When using the Task Manager there is a selection for each Process running. PBXNSIP does not appear in the processes even though it is running. Anyone know why this is the case and what I can do to have it apper?

Still looking for a solution. Can Plug and Play be disabled????

I tried Version 3 and 4 (Windows and Linux) and I don't even see the "Share Mailbox" option. I believe that I looked everywhere and could not find it. Can you tell me where it is????

The WIKI says that you can Share a Mailbox, but there is no provision on the MAILBOX Page to do it. Was it discontinued or is there a newer method? Basically we need to have a single mailbox (General Delivery) that can be accessed by a number of extensions. It would also be nice if they could get a Message Waiting Indication too.

We have a situation where up to 15 extension users dial into their mailbox, listen to one message and then place the call on Hold. A few minutes later they return to the holding line to continue listening to any remaining or new messages and then place it back on Hold again. It appears that this is done so that they don't have to log back into the mailbox every time they want to check messages. Too much work I guess.... Is it possible for the mailbox to release the call after a period of inactivity? We are concerned that this Holdiing method is tying up or unnessarily using system resources.