Vodia PBX
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We are currently running pbxnsip 3.2.0.3144 (Win32).
We use Snom 300 with firmware 7.1.35, this wont work if we have a spanning tree protocol network setup.
Snom say if i upgrade to 7.3.10a or higher it will work ok.
Does anyone know if this firmware works ok with pbxnsip or has anyone had any problems with it?
I would go to version 7.3.14. We are using it, and it seems to be working fine.
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Ok, I finally got to the bottom of this jitter.
We were having a small 10-20% cpu spike about every 15 seconds.
I stopped pbxnsip service and this spike continued.
I did a little research on the service causing the spike (DPC) & found USB devices can cause this. So I unplugged the usb device in the machine and the spike stopped...along with the jitter.
I then upgraded the usb driver for the device and that stopped the spiking. so all is ok now.
Whow! Nice catch!!!
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Ok, I restarted pbxnsip service and it WAS able to set the affinity according to the logfile:
[5] 2009/02/21 16:55:31: Set processor affinity to 2
BUT the jitter remains. It seems approx every 15 sec (very approx) it just cuts out.
-The system has NO load when it happens
-I've tried it with a very simple config, 2 phones, in same switch, no load, still jitter.
-i called sales queue/73 and listened to the moh and roughly every 10-15 sec. it stutters/cuts out for a split second.
-i've tried switching affinity to processor 1 with similar results
If you can easily resolve this, get a Wireshark trace and see if there is anything on the network. Maybe a packet storm every 15 seconds.
I assume it does not make a difference if the call is internal, to the mailbox or external? if the problem also exist for a IVR (mailbox) call we can exclude the jitter is coming from an outside source.
Also, can you check the process list on that computer. Is there anything that has a priority above normal? Only processes above normal can give you that problem.
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I understood what you mentioned after I rechecked the rtp stream.
As you told, PBXNSIP is switching between the codecs during the early media too.
And the user agent cannot accommodate this transition.
Is there any way to change this transition behavior on the PBXNSIP?
Maybe it can switch to second codec which will avoid the transcoding after the 200 OK by locking down to one codec in the SDP .
We will introduce a flag that will lock down the codec one it has been determined. Even at the price of later transcoding.
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When the Hot Desking is enabled, it also affects other extensions in any other domains that have the same extension number.
Oh oh. You are right. Will be fixed in the 3.3 version.
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also, what will trigger the "could not set affinity" log record? i just turned on logging.
If the process does not have the "PROCESS_SET_INFORMATION" permission than the request will fail. Check out http://msdn.microsoft.com/en-us/library/ms684880(VS.85).aspx for more information...
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settings from the target extension:
intercom_enabled!: on
intercom_connect_type!: intercom_connect_type_handsfree
The settings on the phone should not make the big difference. Can you get the SIP INVITE from the phones web trace log?
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[...] i get not appropriate permissions message--soo, maybe that is an issue... [...]
Yea that could be a hint.
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i am having the same problem. The IT admin assigned a local ip to the phone system and has a static IP forwarding to it but i cannot get the Polycoms to register with PLugNPlay and the snom phonen that i registered manually has no audio either way.
the IT admin swears that he has all ports opened to and from the pbx....
It is really a pain if the PBX does not "own" a routable IP address.
You can fiddle with the "IP Routing List" setting to try to undo the algorithm that the firewall does. Very support intensive. You probably need Wireshark to troubleshoot what is going on on the system.
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I am interested in a way to do toll restriction based on a service flag. I have a situation where the phone is publically accessable after business hours and would like to have a way to set the dialplan by service flag so that I can use a toll restricted dial plan only after hours
Whow. I would not know how to do this right now.
Maybe something like the call always goes to an auto attendant, which then sends the call to a calling card. Not very beautiful, as users always have to do two-stage dialling.
I would check if hot desking is an option here. The office-hour user hot-desks to the public phone. When the user is not logged in there, it is just a regular no-permission phone.
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I'm thinking that I have a different problem.
Because I cannot hear the early media too.
And early media is coming before the codec transition.
So, it shouldn't be related with the user agent.
As I told before, I captured the network traffic during my tests.
And I listened the rtp stream with a call analyzer.
There is only a terrible noise on the rtp stream which is coming from the PBXNSIP to the user agent.
Terrible noise is usually a problem with SRTP.
There is a flag for the trunk where the PBX can send only 180 without SDP to the carrier. Some carriers cannot deal with early media!
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Hi since couple days i have a probleme...I lost communication when i'm talking to someone, i think my probleme is this message
PSTN: Number Unobtainable Tone detected on 2 (version: 2.4.2)
i'll past a part of the log... if somewone can help me... thanks
Can you log into the box and check the /etc/init.d/pbxnsip file? is there a "--busy-det"? Take it out and reboot. Check if the problem is still there.
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Hi,
We moved from a Atom1.8g/512mb ram, xp home laptop to a p4 3.4ghz/1gb ram desktop xp pro machine (clean install/nothing else on). Both had 1gb nics. On the new server we now have jitter about about every 1/4 minute. Almost undetectable. (but on vm when people let phone numbers, 1 digit is missing at times)
We have set the affinity to processor 2, but the problem persists.
How long is the delay between the RTP packets? What is the exact time between the problems, 15 seconds?
I assume the process is running with sufficient permissions and there is nothing in the log like "cound not set affinity mask".
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I have 4 Callcentric SLA trunks set up with Snom 360 and 320 phones. Everything works fine except that, if an extension picks up the trunk and hangs up without dialing, the trunk is not released for 60 seconds. If a number is dialed and the extension hangs up, the trunk is released immediately. Does anybody have a work around for this problem?
There is no timer. But when you are "touching" the line (e.g. by listing it in the web interface or by making an outbound call) the line should be released.
We did some changes for internal calls (someone seizes the line, then makes a call to another extension) and maybe within that scope we can periodically check if there is a line that should be released.
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Would it be possible to make an Check BOX option on Manual Service Flags to reset at Midnight along with the other Midnight option to reset hotdesks etc?
This would be selected on a SERVICE FLAG by SERVICE FLAG basis....to reset at midnight.
Good idea.
It brings up the idea that the manual override of the automatic mode must be reviewed again. Because why resetting it at midnight? Maybe you want to reset it at noon, or 3 PM and so on. Then there is a thin line between automatic and manual mode.
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Don't know if this is the right place to start a thread on the FreeBSD port... I couldn't find any started already.
We'll start a FreeBSD Forum.
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what am i missing?
Hmm. Works here... Also on version 7.3.14.
Do you see "No permission for intercom to ..." in the log (level 5)?
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i have the same problem here.
i was never worried about the co lines but also intercom does not work anymore
will write a new post since its not related to this.
The line stays seized if you make an internal call. You can unseize it after the call by pressing the button again or it should also get off after some time automatically (seize timeout). We are putting in that the line gets automatically unseized when the extension makes an internal call.
As for intercom, there is a new settings for the permission to perform intercom. Now you cannot intercom to anyone and turn on the microphone! It was a kind of security leak.
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I tryed to use this feature. The system cut part of destination numbers. Part of CSV :
20090217182847, 202,254495xxxx, 9
202 my extension number
really dialed number was 81097254495xxxx
Looks like too many digits to me. Probably you have to increase the number of digits in the string. For example, change the string "$w$5e$10c$5d" to something like "$w$5e$20c$5d". It is in the setting "cdr_format", check out http://wiki.pbxnsip.com/index.php/Global_Configuration_File for more information.
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We have created a good document on this topic but cannot upload it for lack of space. PLease expand our space beyond the 500K limit and we will post this PDF docs....
No idea how to do that (we are PBX experts, but not forum experts). Maybe just send an email to support@pbxnsip.com and we'll upload it... -
I have a RedHat environment. I'll be happy to try a test build for that environment.
Please try http://www.pbxnsip.com/protect/protect/pbx...hes4-3.3.0.3147.
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Regarding the "Rewrite global numbers" dropdown on the Trunk Settings, I read this in the Wiki:
"When you are using a trunk, you might have to represent the telephone number is a specific format. For example, in the NANPA area, you might want to use 10 or 11 digits to represent a national number. If you are using several trunks, you can represent the same number in different styles depending on the trunk"
The documentation doesn't spell out what this setting really does. Can anyone provide an example of how this setting affects users? Does it just reformat the caller ID? I've tried changing it but haven't seen any effects.
When the domain has the area code set, the PBX convers the incoming numbers into the +xxx format. You see that when you look at the call log from the system admin level. Then when calls are leaving the PBX, it converts the numbers back into human-readable format, or "carrier-readable" format. That's it.
The dial plan also processes the numbers in human-readable format. No more worrying about numbers 1xxxxxxxxxx format or xxxxxxxxxx, or even +1xxxxxxxxxx (it is always xxxxxxxxxx, just like you have it on your cell phone if you are in the USA/Canada).
There is a big difference between the "1" area code and all other area codes. Because the "1" area code means that numbers usually have 10 digits (exceptions being x11, 011 and 555 numbers). The PBX tries to be tolerant against different ways of representing regular 10-digit numbers, e.g. starting eith "1" and having 11 digits or 7-digits for local calls.
For non-"1" area codes, the PBX assumes that international numbers start with 00, national numbers with 0 and local numbers 1-9. Extensions have less than 6 digits.
If you don't want all the magic, you can leave the country code field empty. Then the PBX treats numbers just as strings without trying to be smart.
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Please add to a near future revision the ability to have the PBX use the SIP Replacement list as with SIP/RTP in provisioning so that is uses the replacement address when getting requests mathing that .. this would help provisioning in the real world ..
It does that already for the "IP Routing List". Maybe you can use it instead of the replacement list.
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Alright. Disregard anything I said about the phones ringing after the caller hangs up. That is not the issue I am wondering about. My main question is what is with these fake calls ringing on the phones after someone has been on hold for more than 2 minutes? All the phones are Aastras. An agent group plays music and messages, not a ring back tone. These are not transfers to an extension, these are calls into an agent group. This notion that someone would not wait on hold for more than 2 minutes is ridiculous. Call Microsoft, Dell, or any number of companies and go to the tech queue. All I want is to not have fake phone calls ringing on a phone that is in an agent group and a call is in that agent group for more than 2 minutes. This is my question. 99.9% of my pbx customers asked why do, after 2 minutes, calls that are on hold get disconnected. Now that the 2 minute barrier is not there, now they ask why do extra calls start ringing in.
That sounds like the PBX believes it disconnected the call, the the phones don't. Then when they want to put the call off hold, they send a INVITE (a "Re-INVITE"), but for the PBX that looks like a new call.
Also, callers in the queue are technically (from a SIP point of view) not on hold, they are "waiting" (connected). If all agents are busy, logged out, on DND or whatever, then the caller will stay in the queue, even if it is 40 minutes. The one luring out of the queue and not getting connected for two minutes because there are phones ringing and then hang up on the ringing call, well that call has a problem. It is like an agent presses the cancel button on that incoming call.
What is the PBX supposed to do when a agent rejects a call? That is the question here. From the PBX perspective, there is no difference if a human being pushes a key on the phone or the phone itself rejects the call after a timeout.
Anyway, the solution is to use the redirection feature to escalate the call after some time, maybe one minute. For example, the call can get escalated into another queue (priority queue), to a cell phone, to the boss extension, or even just into a mailbox. But keeping the phone ringing for 40 minutes does not sound like an option to me.
No idea how to do that (we are PBX experts, but not forum experts). Maybe just send an email to
Timed Disconnect
in General Setup
Posted
At the moment my only idea is to set up a second system with a three-minute key and then send the call there.
There is some prepaid stuff coming into the system; that could be used to limit the call duration as well. That would have the benefit that every minute the caller can head a beep and on the last minute three beeps, so that he knows the call will be disconnected soon.