Vodia PBX
-
Posts
11,130 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
I found the solution, It is called Prio from O&K Software. Prio lets me
set the priority of the task and remember the priority at each reboot.
Also it shows you each tasks TCP/IP socket & bandwidth usage. It is
especially helpful since it shows UDP streams as well.
I wish this could be a feature added to a newer release. It would
be useful in both Windows, OSX, Linux and also FreeBSD !
Prio by O&K Software (Free for NonCom & $19.95 Commercial)
Sounds like a good tool. For $ 19.95 sounds like a no brainer!
If you just want to see the sockets, you can also just use netstat.
-
polycom PNP doesnt work.
Now you have to put the username and password into the phone. The good old MAC-based authentication only works when you are in the same LAN...
-
My girlfriend is having a party today and no one can call in. She must love me or maybe she is waiting until afterwards to kill me with the cake knife.
That's pretty serious!
If internal calls work fine, then we must have a problem with the DMZ and/or DNS. This sounds like the original setup had the DMZ set up in a slightly different way. Or maybe the new firewall has a different firmware version, and the forwarding works in a different way. Maybe it also has a logging feature that you can use to find out where the problem is.
Next time, also make a backup of the firewall. Every component can fail.
-
I would like to request a feature to be able to copy a dialplan in the same manner that you can copy a button template for a phone. I find myself spending a whole lot of time creating dialplans that are 90% the same as existing ones.
The head version now contains a feature that allows "global" dial plans that can be shared amongst domains.
But I agree, we need a simple way to copy and paste dial plans.
-
Ive had some random call failed errors on some of our phones and started looking into if we were maxing out our number of active calls we could have at any given time. I dont log every phone all the time, so I cannot look at that, so I usually do a quick glance at the active call status and see only 10-30 calls going on. Looking at the call graph, Im assuming the blue line is number of call legs and the red line is number of calls. If my license has 50 simultaneous calls, is that trunk calls or all calls including one extension to another?
Example is pbxnsip will say 18/40 Calls, and assuming that 36 of the 40 are call legs, the other 4 are ext. to ext. calls, do I have 32 available calls still or 10?
If your setup sends out emails, the PBX will send you an email when a call got rejected because of this. Check if you have an email with the subject "CPU Limit".
-
I did check out the documentation and can't figure out the answer. Main issue is that I can not associate my extension with callcentric incoming phone number alias only works with callcentric account number.
Consider setting the country code for the domain. That usually helps brining the numbers into the right format.
-
I updated the FW to 7.1.39, and so far everything is the same.
As I mentioned earlier..I just updated to pbx v 3.2 from 3.0 a few days back, so I was using this dial plan, as the auto provisioning wasnt a feature previously.
I still havent got the dial plan autoprovisioning working correctly yet. Do you see my dial plan as being a contributor to the problem I am experiencing?
If you set the country code to "1" then you must use 10-digit numbers in the dial plan now (not 11-digit). Then on the trunk you can decide on how you want to present the number to the carrier. You can choose 10-digit, 11-digit, global formats. This could be the problem.
-
Check the status of the setting called "answer_after_policy" on the phone. It should be "idle", not "off".
-
We just turned it off completly. We had "Disconnect call on silence detection" turned off already, I then went in and "SIlence Detection Method" to none on the audiocodes gateway and rebooted them. I will report back.
Do you PnP the snom M3 phones? I guess it would also make sense to turn the silence detection off there as well.
-
Yes, I am able to get my mailbox using *97.
That usually means that the device is working and can make calls...
I also just realized that I can complete a call to a 10 digit number if I first press one of the "Shared line" buttons I have set up. This gives me a dial tone, and I can call anywhere. I used to be able to dial direct from the keypad and it would pick a co line for me.Aha. Interesting. Try upgrading to 7.1.39, that contains additional fixes for the shared line.
Here is the dial plan I am using btw:|^(*[0-9]{2})$|sip:\1@\d|d |^(911)$|sip:\1@\d|d |^(611)$|sip:\1@\d|d |^([2-9]{1}[0-9]{9})$|sip:\1@\d|d |^(1[0-9]{10})$|sip:\1@\d|d
Where do you set that up?! Usually the dial plan should be automatically provisioned!
-
HI;
i am trying to register my phone using PNP but am having an issue where it goes through the entire process but after showing "Processing Config" screen with the IP address it comes back with the error: "Config Eror Code 0x0"
has anyone seen this before? i am using the latest version of pbxnsip.
yes i have tried changing Firmware/Bootrom etc!
Hard to say what the problem was/is (I assume a problem with the matching of MAC to a the user). But it seems that 3.3.0.3156 or later works.
-
Thanks!!! I believe it has fixed the issue. Is Windows 2000 server going to continue to be supported, or will/should I start to look into a larger server to run newer (more bloated) windows software?
As long as the issues with Windows 2000 are not a pain in the neck I think we can continue supporting it. Though the interesting features like IPv6 require something more up to date!
-
I am looking for that as well, so fare no success!
Not the expert here, but I believe that "VLC media player" (see http://www.videolan.org/streaming-features.html) is what you are looking for.
-
OK I just got a Syslog and checked it out. It looks like there is a silent disconnect release message. Here is the log.
Oh can you turn the silence disconnect off on the gateway? That would solve the problem.
-
Can I upgrade form 3.2 to 3.3 by just stopiing the service and then dropping in the the new controller? And which version should I use I have 3.3.0.3147 and 3.3.0.3152
Try http://pbxnsip.com/protect/pbxctrl-3.2.0.3139.exe. Make backup of the working directory before doing this!! If there should be trouble you can revert to the previous version any time.
Do other phones in that network work properly? Is the problem limited to the snom M3? What firmware is on those phones?
-
If I understand this, he is trying to block the cell phones number from displaying as the outbound caller ID, this was supposedly fixed in version 3.2, thus far most ITSP's I tried still display the cell phone.
Oh you mean for the DISA feature when calling from the call phone?
-
Any other suggestions?
Can you call the mailbox (*97)?
-
Application popup: pbxctrl.exe - Entry Point Not Found : The procedure entry point GetAdaptersAddresses could not be located in the dynamic link library iphlpapi.dll.
Obviously Windows 2000 does not support this. Some new functionality is using this. Try http://pbxnsip.com/protect/pbxctrl-3.3.0.3157.exe if it fixes the problem.
-
I wonder if I need to add a "timezones.xml" file to correct this. Gunna play.
Yes the server has the correct time in the logs. and I verified that all phones are set to the same TZ as the server (CST) (-21600)
If the server has the right time then the timezones.xml file is correct. Do you have the change to provision other phones (e.g. snom)? Then we can isolate where the problem is. Maybe the way Polycom wants the switch week is different from what we send.
-
I just started up a new system and put a couple ip450's on it. I changed the sntp server in pbx.xml to point to the pbx. This part works fine, except it is an hour off. I looked in the generated polycom_sip.xml and noticed that the start date for day light savings is blank:
<SNTP tcpIpApp.sntp.resyncPeriod="86400"
tcpIpApp.sntp.address="192.168.10.10"
tcpIpApp.sntp.address.overrideDHCP="0"
tcpIpApp.sntp.gmtOffset="-21600"
tcpIpApp.sntp.gmtOffset.overrideDHCP="1"
tcpIpApp.sntp.daylightSavings.enable="1"
tcpIpApp.sntp.daylightSavings.fixedDayEnable="0"
tcpIpApp.sntp.daylightSavings.start.month="3"
tcpIpApp.sntp.daylightSavings.start.date=""
tcpIpApp.sntp.daylightSavings.start.time="2"
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0"
tcpIpApp.sntp.daylightSavings.stop.month="11"
tcpIpApp.sntp.daylightSavings.stop.date=""
tcpIpApp.sntp.daylightSavings.stop.time="2"
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="0">
</SNTP>
These files are generated, therefore any changes I make are over written by the pbx. Where do I affect this system-wide setting? (it is not in pbx.xlm). I grep'd "daylight" and "ntp" and could not find anything in the xlm files other than what I had already found.
It seems that for some reason the Polycom phones are one week behind with the DST changes. Does the PBX show the right time (e.g. in the log)?
-
i am using a CS410...i have been here for 5 hours already and tried every step...i formatted all file systems reset local config etc...still nothing doing.
pls look at updated logfile after trying http download.
it seems the only issue we are having it the cfg file/license file from pbx to phone.....
should i try downgrading version of pbx?
That log looks fine... The license problem is also okay (just no RTCP-XR reports). I don't think downgrading will solve the problem. Maybe send me a private message with a login (HTTP and SSH preferred) and we can take a look from here.
-
Good morning. Please read my post more carefully. Remote Party ID/xxxx as I posted is also the name of the field in the trunk setup where you configure these options. I never mentioned I was using remote party ID. I am using Preferred and it works fine but it will not respect the ANI of an Extension even if you specify it under ext. config. In order to get this scenario to work of respecting your ANI config you have to use P-Asserted and then it works fine. The problem with P-Asserted is that we loose what we have working under P-Preffered which is having the original caller ID displayed on call forwarded calls.
So to sum up both scenarios work correct on PBXNSIP but my need is not to have one of them working at 1 time. I need both working simulteanously so our current setup will work 100%. How can this be acheived?
It is still not clear to me what the setup is. Maybe you can include a little drawing. No big art, just something scribbled on paper and scan it...
Maybe the problem is that this is not a trunk call? ANI has no effect on calls that are to extensions.
-
already upgraded to 4.0 but still not working..this is a new location and i am registering the phones locally to the pbx.
Hmm. Do you see the generated files in the PBX "generated" directory? Do they make sense (e.g. do you see the account info in the polycom_phone file)?
Can you format the phone's file system and reset the local settings? Sometimes that helps, though it does take a lot of patience to go through all these updates.
-
what do you need?
The INVITE from the phone's log (the one that receives the INVITE) would be interesting!
Number of active calls
in General Setup
Posted
Yes, the red line is the one that is used for the license count. 18/40 means there are 18 calls, and in total there are 40 call legs. You still have 50 - 18 = 32 calls available.