Vodia PBX
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I recently had PBXnSIP in a test environment consume a large amount of cpu time & memory.
The PBX became severly less responsive to phones or trunks causing timeouts. The windows
service was showing it was running and responding. I was unable to go to the webpage to look
at the log or save configuration. It took about 5min but I was able to restart the pbx service
from the windows service manager and recover from the condition.
I found in certain environments I need to have more flexibility in testing and recovery. It would
be nice if there was a tool that would do a SOAP Ping or SIP Ping. Upon a failing condition of a
rule it would do a graceful restart of the pbxnsip service.
Ideally I would have the pbx service being run on another machine in an active or passive failover
or standby. Has anyone done clustering or high availability with pbxnsip under any os?
I would use SNMP to check if the service is still up and responsive.
For Linux, we have written a script that takes care about failover, especially hardware failover. It periodically checks if the service is still there and restarts it if that should not be the case (in Windows, that's the job of the service manager). In Windows, we would need to have a little script that checks after booting up if the virtual IP is configured and if not, grab it and start the service. Probably also no rocket science.
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It takes approximately 6 seconds for the call to start ringing and about 10 seconds for the ringing to stop after hangup on the FXO side. The SIP trunk is immediate.
SIP trunks are great! They treat signalling seperate from the audio, and the other side does not have to guess what the call state is.
There are some carriers sending "Your call has been disconnected. Please hang up... Your call has been disconnected. Please hang up... Your call has been disconnected. Please hang up...". How would the PBX know that it should hang up? No kidding.
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Thanks for the suggestion but I have tried that with no success. I did some tests however and maybe I have a clue about why it's not working. I think that pbxnsip is not serving /favicon.ico except after a successful login. According to wikipedia browsers try to fetch favicon.ico before the user has the chance to login (except MAYBE if the favicon is properly referenced in the html of the page but this doesn't seem to be the case with pbxnsip). If the first attempt fails then they don't try again (at least not until restart of the browser).
Here is what you can do to test it yourselves:
0) place a favicon.ico on some pbxnsip installation
1) Open a browser on a PC that has never visited the pbxnsip installation you'll test with (or trust that you can clear every byte your browser keeps cached)
2) try to open pbxnsip-host-name:8080/favicon.ico -- you should not see the favicon.ico saved in /html
3) login to pbxnsip -- (the favicon should not show up next to the url in your address bar)
4) retry to open pbxnsip-host-name:8080/favicon.ico -- you should now see the favicon.ico saved in /html
BTW I've seen the same "bugy" behavior with images embedded in the emails pbxnsip sends to it's users (e.g. emails about missed calls have a link to /img/save.gif which does not display unless the user is loged in)
Hmm. During the login session we are allowing certain extensions to be loaded ("jpg css gif js"). "ico" was not on that list. I guess that was the mistake. We add that to the list, next version will have it.
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The current time on the server is correct since it works for the other domains in the pbx except for this one.
Are you sure the calls are running in that domain? Maybe be accident they are happening in another domain. Do you see the calls in the active calls in the web interface?
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The call logging does not refresh at all, it keeps showing the same old data (see below):.
Call History
Start From To Duration
2008/12/16 16:11:11 Kantoor 1 (100) 0102535592 14:57
2008/12/16 16:23:09 Kantoor 1 (100) 0102535592
2008/12/16 16:24:34 Kantoor 1 (100) 0102535550 00:37
2008/12/16 16:59:42 Kantoor 1 (100) 0102535550 11:31
2008/12/16 17:11:39 Kantoor 3 (102) 0227591464 02:01
2008/12/16 17:32:17 Kantoor 1 (100) 0227591912 03:21
All the calls that are being made right now, wont be displayed on the call history, i looked for a way to remove this but couldnt find were this call history is being storred, on the pbx server.
What is the current time on the server?
The web page itself does not re-load automatically. You need to load the page again.
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I think you're right. Where can I change these settings? I am from South Africa and it's likely that the system is using different call tone recognition.
It is part of the "PSTN" trunk in the domain. The PBX tone detection is automatic, just looks for a beep beep beep pattern, then disconnects.
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so this will only work for flags in manuall mode?
Lets put it this way: in manual mode it is clear what is going to happen.
The problem with the mix of manual and automatic is that nobody understands what exactly is going on.
I would also like to know if there is a way to change the night service numberMaybe you can combine two flags (or more). For example, one manual and another one automatic. Check the manual flag first, otherwise then check the automatic. This way you have additional flexibility, and it is still understandable. See http://wiki.pbxnsip.com/index.php/Auto_Att...t#Night_Service on how multiple service flags can be mixed.
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i dial 070 which is the service flag, and all i hear is a 2 tone sound and the call gets disconnected. the service flag does NOT get changed
Check if the flag is in manual mode. The sounds are not very convincing (will be better in the next version), the beep is currently the way to tell if the flag is active or inactive. There are actually two beeps, check the audio_moh directory.
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Running 3.2.0.3143 and I can't see any settings to change this. When I go to the PSTN Gateway settings. I can only see the attached fields and no mention of things such as "Detect Busy Tone" mentioned here http://wiki.pbxnsip.com/index.php/Installi...P-PBX_Appliance.
I would turn polarity change off, this is usually just an unneccessary source of problems.
So you mean, when the call comes in on FXO, the PBX calls an extension, but when you hang up on the FXO side, the call keeps ringing?
Do you have "Requires busy tone detection" set to "yes" in the PSTN trunk in the domain? Looks like the carrier sends a busy tone, but the PBX does not recognize it.
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i am not sure how the ani can be used when we need the digits in the dialed digits (dnis)
Ehhh that is true.
Did you try something like this in the dial plan replacement:
sip:1234\1@\r;user=phone;other-arg=lalala
The "\1" is the match from the pattern, the \r is the domain name. This will go into the Request-URI of the request. Is that okay? The To-Header should usually not be used for routing purposes according to the SIP RFC.
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I'd like to change the favicon displayed for the web pages of pbxnsip. Is there a way to change the default one?
You can put your own into the html directory (call it "favicon.ico"). But beware - the browser stays with the old one, even if you press F5. At least IE seems to keep it for a long long time.
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from my original post:
i saw the prefix field under trunks but after reading up on it i see that its something entirely different
here is the note on prefix from the url that i reviewed prior to the post:
If that ANI is not set, the PBX checks if there is a Prefix set in the trunk. If this is the case, it puts the prefix in front of the primary account number and uses that as ANI. This is very useful in cases when a trunk deals with a range of numbers (typical outside of the NANPA area, e.g. Europe). The "extension" number is just put behind the prefix.
here is what i need:
domain xyz tech prefix is 478325
user dials 12134447777
pbxnsip sends 47832512134447777
like an auto insert to the outbound digits
since all traffic from pbxnsip gets sent from the same ip address we need to identify the domain making the call and can not think of any other way to do it
can this be done outside of the dial plan?
Oh, did you check out ICID (RFC 3455)? This is a simple method of tagging the request for outbound calls so that the upstream provider can tell where the call came from.
The prefix could also be a way for you. However, you must use ANI.
However, I would rather use common prefix in the ANI to have the same effect. Like the "area code" in the DID number.
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The problem is this line:
[5] 20090330161234: SIP Tr udp:196.28.14.74:5060:It means the PBX sends the Re-INVITE to the gateway (repeatedly), but does not receive any response. The gateway should send anything, even if it does not like the Re-INVITE. Do you see something in the log of the gateway? Maybe it tries to send something, but to another address (not the PBX)?
USER-AGENT: RTCC/3.0.0.0Maybe there is a newer version available. It could be the gateway does not like the port 0 in the SDP (the PBX just relays that information).
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Do I need an new licence?
No. Let us check what is the problem.
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I have a few clients that are set up with a manuall service flag, i was wondering if there is any way to call into the system from anywhere and change the status of the service flag
I would also like to know if there is a way to change the night service number i have set in the Agent group
i understand it might be a security issue that anyone calling that customer will change things around when tryign to figure out someones extension etc, so maybe have a Special "Account Type" that will be for administrative modes and have a DID go to that Account and require to enter a Pin number, maybe even build a white list of only which number can call in and access that special account
i personally think this will be a very strong tool
Should be possible. When you call from a cell phone into an auto attendant, you should have the option to make an internal call - which can also be a service flag. I am not 100 % sure if it will work (did not try it out), but that is worth a try.
If you are not calling from a cell phone (just any inbound call), you could send the call to a calling card account, from which you can make an internal call. Same story.
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Id like to have an option of pressing 2 to send to voice mail, pr pressing 3 to transfer to an extension, etc
Makes sense. Even the transfer could be possible, as soon as the called party presses 3 the call forking can stop and the PBX patiently waits until the extension has been entered.
Will be a 4.0 version feature, though!
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outside of manually inserting a tech (leading digits, ex: 4387652 in front of 1npanxxyyyy) prefix in to each and every dial plan entry for each domain, is there a way to specify a tech prefix per trunk/domain ?
currently when we or any other itsp routes calls to a multi domain pbxnsip server the call has to be sent to the fqdn url of the domain and not the ip address as only localhost domain resources are available when the call comes in to an ip address
so on outbound traffic the pbxnsip server will send to the ip addresss of the itsp and with the ip address of the pbxnsip server and not the url of the domain
so the only way we can think of identifying domains is to use a tech prefix function per domain so they can share the same public ip address for outbound traffic
so in this config the billing system will bill for two trunk ids. one for origination (url based) and one for termination (static ip with tech prefix)
i cant think of any other way to do this but need to know if there is a clean way to specify a tech prefix per trunk per domain
i saw the prefix field under trunks but after reading up on it i see that its something entirely different
If you can register the trunk then inbound trunk identification is very easy for the PBX. It just uses the "line" parameter with a trunk identifier in it.
If not things indeed get tricky. Of course the IP address of the PBX is a very poor indicator where the call should go.
We found that using a DID number is the most natural way of finding out where the call should go. This is something that almost everybody understands "naturally", as it reflects the way people are calling resources in the telephone world. Even if it is a dummy DID (which cannot called from old grandma's phone) it still serves its purpose (note that there are more telephone numbers in North America than IP addresses in the world!). By adding the "ANI" for outbound presentation, and by having the telephone number as alias name of the account for inbound search, this is a easy way to do inbound and outbound mapping. You can even setup a global trunk that takes care about sending the call into the right domain.
In 3.3 we also added the possibility to have a global dial plan - which makes it more convenient to set up complex dial plans only once.
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I would like to know if it is possible and how we can set up call forward to dial an external number that include a pause in the number.
something like a cell phone
1 555 555-1212T101
The PBX does not support that. We leave that job to the PSTN gateway, some of them support two-stage dialling.
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Even when I place the access number added to the callback extension, no dtmf tones
If the other side uses only G.729 without and out of band DTMF (RFC2833/4733), DTMF is not possible as G.729 distorts the tones, so that the detection becomes impossible.
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It is indeed a model with a single port on the back. Unfortunately, it was an eBay purchase, so I won't be able to swap it for a black one. Do you know whether this a recoverable situation, or do I now have a very expensive brick?
Talk to sales of pbxnsip, chances are you get a replacement with a black box.
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Has this provisioning method been completely disabled from outside the lan?
http://pbx.pbxnsip.com:9012/provisioning/s...0041324006D.htm
What is the correct method to provision from outside the LAN, and obtain all the phone settings.
I tried putting the MAC in the extension, and using http://pbx.pbxnsip.com:9012, the phone logs showed some redirects, time settings being applied however thats about it, it did not register.
3.3.0.3165 (Win32)
In 3.3, there is a change in the way you need to provision snom phones. The new link must be in the form http://ip:port/provisioning/snom300.htm (where the snom300 needs to be replaced with the actual phone model). We'll update the Wiki as soon as 3.3 is publically available.
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When I "copy" a message to another mailbox via using key "6" I am precluded from continuing with my call and listening to my remaining messages as my phone sorta locks up. Is this normal? Thanks.
What do you mean by locking up? Do you have to reboot the phone? Or does the PBX simply do not react to DTMF any more? Maybe there is a audio prompt misssing, so that the playback stops and the PBX waits for more input.
i have also had trouble when pressing other function keys during a voicemail that the audio becomes garbled until i hang up and call back.Are there any other cases when the playback gets garbled? When listening to voicemail, "1" and "3" are backward and fast forward; which is always a little bit garbled. Maybe that is the problem?
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It was this part of the update process that the box has failed after...
Updating the MSP
Once that is done it is necessary to repeat the step and use the msp update file to update the DSP firmware. The system will now be running version 3.0 and go to system software update tab and load the update-msp.tgz via browse, then save, then restart and the system is running the latest MSP.
This is part of the instructions at http://wiki.pbxnsip.com/index.php/Installi...P-PBX_Appliance
I do hope that following these instructions hasn't trashed the device, as that's a pretty grim introduction to a product.
Where did you get the white box? So it has only one Ethernet port on the back? Maybe you can try to swap it into a black box with two ports.
The problem with that box is that if you apply the software update, it will get tricky to get back to the back, as the IP config probably got screwed up.
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The exchange server is running of sp1 ..
does this effect the ivr
I have no idea. But usually it is a good idea to use the latest service pack.
how to setup trunk to send a tech prefix to carrier
in Trunk Setup
Posted
What does your dial plan look like?! Too many entries are not good anyway... Check out the regular expressions, they can make your life easier.
Also, did you try the prefix? If there is no trunk ANI and also no extension ANI, the PBX will put the prefix in front of the user name. For example, if you are calling from username 123 and the prefix is 123456, then the PBX will make 123456123 out of it.