Vodia PBX
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Yeah I think it could be possible even our old ISDN pbx could do it... so why a modern VoIP system can't?
The problem is the failover in the dial plan. Not a big problem, but we don't want to bring new features into the 3.1 release at this point.
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one of Fonality's selling point is agents with cell phones. obviously you need to press 1 to accept the call.. but it is a cool feature.
Well, maybe then they should also start support calling two registered devices on one extension...
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Well the support only WMA or MP3 or Windows Media Streaming stuff.. www.shoutcast.com shows a list of known shoutcast servers.
Well, then you need something that converts this into plain audio. We ran a MP3 player that supported Internet radio for some time and had to find out that this thing had a memory leak that eventually brought the host down...
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When pressing the "Record" button using a Snom phone the PBX does not email the recording anymore. The pbx does email voicemails fine so the email part is not the problem.
This was working before.
The traces do show the Snom phone sending the Info message with the record=on option and PBXnsip responds with 200 OK.
Seems like a bug to me, please check
Probably a problem with the license. In 3.0 you need the recording license for this. In 3.1 we split this up into a "full" recording license and a adhoc-recording license. The adhoc would be for the user-initiated recording that goes into the user's mailbox.
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Im having a similar issue.
i have polycom phones and i am giving out DHCP directly from my Windows Server. the problem is that the phones dont have access to the internet..what is the best way to get these phones to pick up the time without the polycoms overwriting what i put in as the time server?
<PHONE_CONFIG>
<OVERRIDES tcpIpApp.sntp.daylightSavings.stop.date="1" tcpIpApp.sntp.daylightSavings.start.date="1" tcpIpApp.sntp.address="http://10.0.10.16"/>
You can set the address of the time server in the pnp.xml file in the setting "ntp_host". See http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to change that setting without having to change the pnp.xml file or restarting the server.
In the new versions of the PBX we will include a small NTP server that can also be used instead of a public NTP server. This way we get rid of such problems.
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Hmm, is that the right bootrom? I remember it should be something like 4.x for SIP version 3.1.1 of Polycom?
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Is it possible to use a SHOUTcast or ICEcast or some other stream media as MoH? or select a directory per domain wich contains wave files to play? (so for each domain other files can be selected and played randomly or something...).
What format are they supporting? Do they support RTP? What codec? G.711? Then that should be possible. You can also have more than just one RTP MoH source, and then you can assign it to different domains.
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I have the registration refresh at 60 seconds as i am using TCP i am not getting this error once i get it in blocks of around 20 lines!
is this an issue that needs to be fixed?
No, not really neccessary to fix it. There are two parts in the PBX, a lower part doing the SIP work and an upper part, doing the application logic. It can happen that they get a little bit out of sync - for example the upper part responses to a registration request while the lower part already removed it. In such a case the lower part would complain that the registration parameters cannot be applied. Such a case gets more probably when the upper part takes a long time to process requests. I would only be concerned about this if the overall CPU load gets too high and the system is on the edge.
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Thanks for this level of detail; however I'm having trouble just finding where the CDR is. The only reference I can find is the Wiki one; but in terms of finding the tool, activiating it and using it we're a bit lost.
What are the first steps for practical application of the CDR tool?
The "CDR Tool" is/was a collection of PHP scripts that work with mySQL. IMHO it went too much into "spaghetti". If you can wait a little, then I would say the option to use direct writing into mySQL sounds more suitable to me...
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Do you mean, that i should redirect all calls to the same number?
Or do you mean that i should use separate extension for each destination number, that is not a local extension? (It means that customer will have to pay for each destination number, which is being routed to different sip server.)
Yes, the seperate extension will solve the problem. Yes, this will require more licenses (think about the margin that you make, hehe). If they are buying another PBX license for the branch office, maybe it is possible to make a package deal, so that the costs are not exploding.
What i'm talking about is quite standard task: routing of incoming calls between several offices...Routing is making a decision (depending on dialed number) if incoming call should be sent to local extension (phone, vm, aa, acd queue... any type of extension) or it should be sent to another sip trunk (trunk selection should also be possible based on dialed number).
Without transit calls routing pbxnsip just does not suite for customers with several pbxes.
So i have to ask again: can pbxnsip route transit calls?
If not, may be it's worth planning this feature for the nearest development?
Today I would technically solve the problem with just more extensions and do a deal with the licenses.
We have to check how much side effects it would be if we allow the trunk destination to be either an account or an external number.
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They have an enterprise class router, there is no NAT involved, most complaints of this problem have been after business hours, workers calling in to check their voice mail, there were no other active calls on the server at the same time.
I had it a few times myself when I called in, it seems to forward to the same person usually, I once got someone's voicemail, however it disconnected before I was able to hear their complete phone number. It may be a coincidence that every one who complained about this problem was using Sprint cell phones.
Whow maybe Sprint is now also using VoIP and they are mixing ports up? Haha, just joking.
Another easy thing to check would be that you have enough RTP ports. Having 1000 RTP ports does not hurt, even if you have far less calls.
If the problem does not go away, you might have to start Wireshark (maybe rotation mode) on the system. This will involve some data mining, but then the root of the problem will become obvious.
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I'll try this tonight as it requires the services to be restarted.
Probably no restart necessary!
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Where is the CDR tool? And is there a place where you can specify what's recorded in the daily emails?
On the agent group there is an option to send the daily CDR report to an email address. Can this be sent to more than one email address and if so what kind of seperater needs to be used to acheive this.
Regarding the CDR you can do the following things:
- Keep them on the PBX
- Use the SOAP method to push the CDR out. That requires that you run a SOAP-enabled HTTP server somewhere in your network that parses the messages and then processes them (possibly put them into a local DB).
- Use the "Simple CDR" method. This essentially avoids the relatively complicated SOAP method. Instead of writing and setting up your own software, you can use existing software like Metropolis that will process the CDR and generate the reports that you like to have.
- You can receive the CDR in a daily email. This can be done on per-domain basis or per-extension basis.
I know that the CDR topic is important. We are thinking about two more options: (1) Appending the Simple CDR to a file just like we do with the log file. The file name may include the day name, so that you get one file per day. (2) Writing the CDR natively to a SQL database (probably mySQL first).
- Keep them on the PBX
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Hmm. If the caller calls a resource on the PBX (say an extension, or an auto attendant, or something else), the PBX is able to involve outbound calls. For example, when calling an extension, the PBX can fork the call also to a phone number. Or when calling an auto attendant, the call can also be redirected to an external number. So far, that seems to make most of the users happy.
The only "problem" we had so far was Microsoft Exchange and it's click to dial feature. Because then the call comes in on the Exchange trunk and it is supposed to go out on another trunk. That problem we did solve with the "Accept Redirect" flag and the "Assume call comes from extension" setting. In this case, all calls from that trunk are redirected to an outgoing number - even if the call might go to a local resource.
I believe your problem can be solved just by using the "redirect all" feature of the PBX. Just use a local extension and then redirect all calls to the PSTN number.
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Is there a description available of the Applicable Tables and respective columns desribed on http://wiki.pbxnsip.com/index.php?title=Ac...on=7#extensions .
I am looking for the information which each described item exactly means.
The items are the internal representation of the functionality described in the Wiki. There is no extra description on how these items are stored in the database. And IMHO you should not touch it, as it is "subject to change without notice"...
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We seem to have the same problem, did you find any solution for this ?
I tried versions 3.0.0.2998 (win32) and 3.0.1.3023 (win32) but calling out became impossible.
I keep getting "Unauthorized" when i try to dial out.
http://forum.pbxnsip.com/index.php?showtopic=1249
Tried all the settings in this post but still no calling possible.
Currently running 2.1.14.2498 (Win32)
Do you have the account set and/or username in the trunk? What trunk type are you using?
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1) it happens all day to different extensions...or every time i reboot the server.
Okay...
2) what do you mean by "the carrier is rotating IP addresses"?Some service providers give you an IP address only for one day. This is because they just running out of IPv4 addresses. When they change the IP address there is a "hickup" in the network that would explain why you get the warning from the PBX.
3) we have a static public ip address and the phones get their local IP using windows DHCP server with option 66 on the phone server?That is fine.
4) right now the internet access is coming into the phone server through a SonicWall router, but the phone calls are going out using a PRI card from Sangoma. do you think it would make life simpler to just switch out the sonicwall for a basic Linksys router?I believe SonicWall is a pretty reliable device. So I would think that the refresh time is too short. Check your admin settings for "UDP NAT Refresh" and consider making it shorter. Also check the settings of the Firewall - the SonicWall has a lot of them and maybe there is a setting where you specify how long to keep a NAT binding alive.
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We have strange issue with one client running 3.0.1.3023 (Win32), they use Broadvox as their ITSP. several times maybe once or twice a day people who try calling in from Sprint cell phones do not reach the autoattendant, a total stranger picks up the phone and would not identify himself.
The call logs shows the call came in to the auto attendant, I don't see the call being forwarded elsewhere, in the log files.
Does it sound like someone hacking in to the system and intercepting calls?
I attached a log of 1 of these calls, I can't reproduce on demand however if I try enough times it happens, this just started over the last few weeks.
This is usually a sign that there is a port conflict with RTP. For some reason, two calls point to the same IP port. This might be caused by the router (if it is buggy) or maybe the carrier has a problem there (less probable I would say). Some routers have only 32 NAT entries, and if you have too many NAT bindings you may have an effect like this.
How to troubleshoot this? Difficult. If you have a cheap router, I would just try another router from another company and see if that changes anything.
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Does PBXnSIP support transit calls?
If yes, how do i setup it?
If not, is it planned for future development?
Can you elaborate on what a transit call is?
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I KEEP HAVING THIS MESSAGE COMING UP IN THE LOG...IS THIS SOMETHING TO BE CONCERNED ABOUT?
Source address for sip:211@localhost has changed to tcp:10.0.10.111:34300
If it happens every couple of minutes: Yes, either your refresh time for NAT is too long or you just have a router there that is not suitable for VoIP.
If it happens once per day: The carrier is rotating IP addresses. There is nothing that you can do apart from mailing them and asking them when you can have a IPv6 address...
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How about importing users via LDAP from exchange or other ldap server?
It would be convinient feature for new deployments...
At the moment you would have to go through CSV. There is a bulk import feature in the web interface, the last dropdown on "create account".
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Hi. We have put the default time server in this field but after the phone reboots it goes back to our server for some reason. Can anyone think of why this is happening?
The version that you have probably has the NTP time server set (in the admin/ports page). Just clear it and then it should use the default time server again.
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So the question is: is it a proper way to route incoming calls?
The handling of calls coming from a trunk were designed to send them to internal destinations. If we are starting to send them to external destinations we get into new areas. The problem occured for Microsoft Exchange and OCS; so we introduced some ad'ons to make this possible. But I would not call that a "proper way" to route incoming calls.
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Here is what it says after only 11 hours of uptime:
0 11:30:54 (105MB/2046MB 19% 23248704-0) WAV cache: 0
-11384 0 200 Ok
-11383 0 BYE sip:12482522626@63.214.44.25
-11382 0 0
-11381 0 0
-11380 0 0
-11378 0 BYE sip:19529083150@64.192.112.13
-11377 0 200 Ok
-11376 0 REGISTER sip:sip.gafachi.com
-11375 1 REGISTER sip:sip.gafachi.com
-11373 0 200 Ok
-11372 0 0
-11371 0 0
-11369 0 0
-11367 0 0
-11366 0 0
[some other stuff deleted]
Aha. I think the problem must be some keep-alive traffic. ... can we have a Wireshark :blush: ?
Broadvox Static Trunk Setup
in Trunk Setup
Posted
Well, in gateway mode you can use the "domain" name to present something else than "localhost" (for example, you can put your public IP there). Always use the outbound proxy! Broadvox also changed their software a lot, there is no need to turn UUID off now any more.
Registration mode has the advantage that you can register easily; but the caller-ID presentation is more difficult. Therefore I would tend to prefer the gateway mode. You can switch the trunk in the PBX between gateway and registration mode either during the setup or later in the select field when editing the trunk.
BTW I think it is awesome that Broadvox supports both register mode and gateway mode.
I see you like to good old stuff. Consider moving to the latest (maybe first with a 3-minute demo key, get it here: http://www.pbxnsip.com/trial). A lot of new feature and improvements over the last couple of years!