Vodia PBX
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I've tried to use the Outbound proxy IP address instead of the DNS name, but the problem persist.
My PBXnSIP server is connected to the internet by using a NAT router. The PBXnSIP server is reachable form the internet on TCP and UDP port 5060.
The PBX sends out a REGISTER package.
Hmm. Okay at least we know it is not a DNS problem...
If the REGISTER messages are being sent out, then there must be something in the path between the PBX and the provider that causes this problem. At this point, it seems like you need to use the "divide-and-conquer" method to isolate the problem (found an interesting article at http://en.wikipedia.org/wiki/Divide_and_conquer_algorithm).
Apart from that, it can be practically anything (see http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems). We have seen cases where the NAT router goes belly up.
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... is there any way to test this ?
You can use Wireshark to capture the traffic to the service provider. Wireshark can a convenient way of extracting the audio and write it to a file, and you can play it back with an audio tool.
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I have some issues with my trunks. I've got two trunks, one is setup for VOIP Buster and the other one for Budget Phone (Dutch VOIP provider). When I click on the register link, the trunk seems to register without any problems. I see the status "200 OK (Refresh interval 20 seconds)". Everything works normally, I can call in- and outside. But after some seconds the trunk status change to "408 Request Timeout (Registration failed, retry after 60 seconds)".
That is indeed strange. Anything from the log? Do you see that the PBX sends the REGISTER out? Just a wild idea, try using IP address as outbound proxy, maybe the is some tricky problem with DNS.
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I like the XML button on the snom. but would like more XML
such as the ability to replace the voicemail button with a virtual voicemail XML app.
maybe an application to see queue stats..
a button to see parked calls.
Yea we already put in a menu button so that you can change DND and cell phone from the phone. That's a start. I like the idea with the parked calls.
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We are using the call limit on a domain by domain basis to limit the call paths to a given domain based on what they purchase. That given we scale the box accordingly.
However what complaint we have hear is when the call limit is reached they get denied, obviously and the customer wants the call to go to a voicemail. Its a feature the RBOC/CLEC's can give but i cant.
The other problem is that many customers want to limit the number of calls because their Internet connection can have only so and so many calls. "Squeezing in" another mailbox call does no good here.
And I would not stopping my customer making phone calls that generate revenue... So if the bandwidth and CPU capacity is right, I would not limit the number of calls at all!
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I have only one 550IP in our enviroment, the phone registers and can be used, however, it always thinks there is a new vm, has the light blinking and the studder beep even when there are no new or even messages at all. It also tabulates missed calls even when you clear them. Any help woudl be appreciated.
Does that stabilize after some time? What after a reboot?
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I am using 2.1.12.2489 (win32) and still have problems with bad quality moh, even de default music.
Voice quality of the Agent Group is fine
The music that fades in and out is terribly distorted.
I wonder if i am the only one here having the problems with the default music as well
Are you using comressing codecs? Like GSM or G.729A?
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Let me clarify my understanding of your instructions:
For a point to point ISDN Configuration (PBX exchange) I would normally freely associate to an extention (ISDN Point to Point) (2-5 digit City Code) + (6-7 Main trunk number) + (2-3 digit extension).
That I grasp.
For a Mulitipoint to Point (Multiple Line Exchange) I should set up the Lines with unique ANIs. That does not sound right to me. If I have one BRI that allows only 2 simultaneous calls over the ISDN line, then the outgoing line could be in use by a number of identified numbers. I need to associate the incoming call to the specific extension. It would seem to me that setting up the tel alias is quite enough to allow the call to be directed to the proper extension.
The point I think you were making was about another point than I was trying to clarify. I wanted to determine if it is normal to set up 2 Trunks for every BRI on an ISDN line, or is the system smart enough to use the Line B without being programmed. My question about falling over was framed to clarifiy if one ISDN is busy, will the second caller calling over the ISDN line get a busy if no 2nd Trunk is listed. I was having symptoms of such problems that led me to install a second trunk.
I have to admit I dont 100 % understand what you mean by "multipoint" and "point to point". I know that in ISDN a "line" can have different From/To identities (because the signalling is outside of the "line"). For example you can have ten MSN, but only two calls at the same time.
I would just set up one trunk for one BRI, and then have two CO-lines on that trunk. The CO-lines are not bound to specific MSN, so there should be no problem. The pbxnsip "trunk" can have more than one call as well, pretty much like the ISDN model.
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What is the purpose of the ANI in agent groups?
I though maybe if i logged into the queue and made an outbound call it would be the ANI from the agent group... I was wrong.
Well, it is a technical thing. Everytime when there is a dial plan, there must also be a ANI. Because the AG probably always redirects calls, it will probably not use the ANI. But maybe one day we have an option that says "keep the ANI", and then the ANI becomes important.
Also for the network identity, the ANI is already important today. When redirecting calls, the AG gets charged for the call.
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So, I can assume that qhen the "official" license arrives the G729 will work...
Thank you for the answer, unfortunately I didn't find this info on any document and it's a bit "heavy" using G711 on a ADSL trunk
Yes.
G.726 or GSM can be an alternative. They don't compress as much as G.729A, but the difference is only 13.2 - 8 = 5.2 kBit/s (GSM); considering a packet overhead somewhere in the 20 kBit/s area, that should be a viable option.
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I think it would be nice in an hosted environment to set the domain call limit to say 4. and then have an option of the 5th call comes in to force a redirect to a voice mail (or whatever)
Right now if they hit the limit they get denied, but if we could allow this somehow we could charge the user for a voice mail box and not deny the call. Kind of like offering voice mail like Qwest does.
Well, the problem is that voicemail might be even more CPU expensive than just running the call... The call limit was introduced to make sure that there is enough CPU for everyone.
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What has been your experience with setting up a system with a Cable provider and Cable modems? Are there problems or not?
In USA it is practically the only choice. Works surprisingly well! But of course the big question is how to ensure QoS.
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I have installed the lasted 3.1 build. After that I've configured the trunk settings (Rewrite global numbers: "+" style).
And also change the country code to 31 in the domain settings.
But it didn't solve my problem. Inbound call are still in the wrong format. (0351234567 instead of +31351234567)
Do I need to change the trunk settings of my VOIP provider or the one to the OCS mediation server?
Hmmm. Seems that this did not completely solve the problem. I guess we have to try this out here as well...
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I'm running Windows Server 2003 (32bit)
Check out http://pbxnsip.com/protect/. See http://wiki.pbxnsip.com/index.php/Installi...#Manual_Upgrade on how to upgrade. In this case, you should make a backup before doing this.
In the trunk, there is a setting that defines how to present the number. Choose the "+" style. You must also choose a country code in the domain so that the PBX knows how to interpret numbers.
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I'm having some trouble to translate incoming calls to the e.164 format.
My PBXnSIP installation is connect to OCS 2007 server with a mediation server in between.
I've created a PBXnSIP trunk to a Dutch VOIP provider (Budget Phone). The trunk is configured to forward all the calls to a user extension. (Send call to extension: 600).
When I call my Budget Phone number from a PSTN phone the trunk will accept the call and forward it to extension 600.
My communicator client will ring and is show the caller-id of the incoming phone call. This displayed number is not compatible with the e.164 standard. With means OCS is unable to do a reverse number lookup in AD or Outlook.
Can somebody tell me how to translate incoming calls to a e.164 standard before they are forwarded to OCS??
That topic is pretty well addressed in 3.1. If you like, you have a preview image. What OS?
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Hallo, I'm trying to use a G729 codec on a trunk, Eyebeam connected directly to the itsp works well in G711 and G729, The trunks with no codec-preference works well also, when I force the "18" Codec I get the busy tone.
I'm using a trial version, will I have to buy the "PBX-LRC" packet?
I thought I had to only to connect to G729 Phones (or multi-codec ones, to avoid translittering going to a trunk like this)
On a trial version, you cannot use G.729. It is not because of us, it is because of the license/patents associated with that codec. There is no "free" with this codec, for even free try out.
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I am wanting to understand how a Point to Mulitpoint ISDN configuration should be set up differently from a Point to Point ISDN configuration. I want to avoid mismatching the incoming calls to specific devices when it is a Point to Multi-Point ISDN phone line.
It seems to me that a Point to Point Configuration: where the provider gives a main number and an additional range of numbers which may be addressed (usually sequential) is fairly straight forward. Enter the Trunk--->assign the extention within the trunk and one is more or less finishted because the telalias can identifiy the inbound based on both assignments.
With a Basis rate configuration that is Point to Multipoint the provider usually gives a list of up to 20 numbers, which most of the time are not sequential, and usually are not logically related. Because they come in over the common ISDN trunk and the numbers are assigned by the provider I have had some small problems with the setup.
By chance I found that one must set a trunk for as many ISDN Channels as one has. Is that correct? 1 BRI = 2 SiP Gateway Trunks, so that with 2 BRIs one would require 4 SIP Gateway Trunks.
The telalias is then used to provide number assignment (MSN) to the extentions. Is that correct?
Otherwise, what is the best method for assigning these numbers when one has several aliases that must be linked to a limited number of extensions (physical telephones)?
It is my understanding that with a Point to Point Trunk setup one may arrange for Fall Over to the next Trunk. IS that also possible with a Point to Multi-Point Trunk setup?
You can use the "Prefix" setting for outbound trunks. Then the PBX will just append the extension number to the prefix and uses that as the "ANI" (subscriber number, MSN).
Inbound you can use a expression to extract the extension. There is a example in http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk where you can see how to extract the last few digits from a incoming number.
With this, you don't need to specify any alias. All you need to do is to configure your trunk correctly.
Failover is a topic that is not related to inbound traffic; and for outbound traffic it is not related to the caller-ID representation.
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Thanks. I don't have the equipment to check the FXO with any certainty but I got a few phones and ran tests with each line and the crosstalk does not occur where a line (any and each) is removed from the PBX and used for incoming or outgoing calls. The noise and crosstalk are much more pronounced on the inside than the outside. If on a call, and another call comes in, the ringing of the other call comes through as a low humm ticking kind of sound, then a very brief noise sounding like a modem handshake and then the voice of the auto attendant.
I've had to revert to the old phone system as no one was able to even have a full conversation without a full disconnect or crosstalk making conversation impossible. As for demarkation, we're in an office building, I'm coming right off the panel less than a foot with standard phone lines to the PBX, the floor demarkation point is only about 30 feet away.
Not sure at this point if the box is a dud or I have it horribly misconfigured, I have not deviated a whole lot from the default configuration. I can't run valid tests now until before/after hours.
I think the core problem here is the crosstalk. If we have such a strong crosstalk, then obviously things like DTMF detection become instable and unpredictable. If we are able to solve that problem, then maybe the other problems also go away with this.
An internal shortcut between the FXO connectors in the PBX would explain this (that would be clearly a RMA case), but such a production mistake should be extremly rare. I would say if you don't find anything with the crosstalk, send the device back and we have to check if there was anything wrong with this specific device.
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What is the cause of a caller to the pbx not receiving a tone on their side that the phone on the pbx side is ringing? Inbound callers are experiencing a long silence until the mailbox picks up.
Which files need to be installed and in which file ordeners to correct this problem?
I assume that you have audio_en, audio_xx (e.g. audio_de) and audio_moh installed?
Some carriers cannot deal with early media, and then you need to set "Ringback" to "Message 180" on the trunk.
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on the old version i could force a certain ANI for a trunk group. even if the tel: was available.
this doesn't seem to be the case anymore. Is there any way to force ANI on a certain trunk?
I have a customer who sends 8001111111 as their caller ID, a toll free number.
However when they call toll free number with a toll free ANI the call gets rejected. If I could build a trunk for the these calls and force the ANI to a local number i would be just fine..........
Calling a toll-free number from a toll-free number. It would be a great money saver if that would work!
At the moment, there is no easy workaround. I think we need to make it possible to have several ANI numbers and then limit them for trunks. For example "8001111111 Trunk1:9787462777".
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Yes, the dtmf_gain is set to 512.
The first test was on IVR, now we test on AutoAttendant.
There is always the problem.
Well, the PBX is not directly involved in the FAX. Regarding FAX, it does only two things. First, it detects the FAX tone. This is like a DTMF tone, and this seems to work (at least using the file that you sent us). Second is the switch to T.38. This step is neccessary to make sure that FAX data does not get lost. If you keep the calls in the LAN then there is usually no problem. However if you are using a service provider in the Internet, and that service provider does not support T.38, then it is almost impossible to send FAX. There are always a few packets that get lost when sending FAX over the Internet, and that screws up FAX.
One more thing comes to my mind: Did you set "inband detection" (admin settings) to "on"?
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We've just installed a CS410 - 3.0.1.3023 (Linux) and we are having several issues:
1. External caller ID will not come through to any phones but does eventually show up in the logs as below:
2007/12/31 19:40:59 Anonymous (anonymous@localhost) Jeff *** (227@localhost) 00:05
2007/12/31 19:41:32 JEFF *** (613*******@localhost) CO4 00:19
2. If someone is on a call then the system simply won't pick up consistently, we have 3 PSTN lines connected. (in fxo1 - 3 so the CO4 thing above is confusing)
That sounds like there is a problem on the analog side. The only explanation that comes to my mind is that the FXO believes that the caller disconnected the call after the first ring and then calls again. What line is it? Did you already use a tool for measuring out the analog line? Long lines can be a big problem for stable operation (that's why many people move to digital lines).
3. Certain companies that use menus (like press 1 for english) are not picking up the user input (I think I may have fixed that with the gain control)Again, if the DTMF seems to be on the edge, it is an indication that the PBX has a problem with the line.
4. If someone's on a call, they hear EVERYTHING from any other line, the ring, the auto attendant pick up and the voices, both ends of the call hear it as clear as day. These same lines were hooked into a Nortel switch for years with no issues.Phones are eyeBeam 1.5.19.2 and Pollycom Soundpoint 330.
Whow. That almost sounds like there was something with the patch panel.
Anyway, all those problem relate to FXO. I would suggest using a tool for measuring out the FXO and/or using a signal amplifier to get the quality under control
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I would like to know who has a successful VOIP supplier in Germany and a link if you have it. I am trying to work out my problems but if that is unsuccessful I will need alternatives.
Toplink works, IMHO their SDSL offer with QoS is great.
I remember QSC also works. And I remember ClaraNet also works.
This list is definitevely not complete. If I forgot someone, please just post. Ideally including some configuration tips!
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Why is the mac installer zupplied in a zip file (zip isn't a mac file format ).
I find that often, like in this case, it doesn't come through properly and therefore be uninstallable.
What would be better? TGZ?
REQ: hosted feature
in Product/Site Comments
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So we checked what that means to the code. At the point where the PBX makes the decision it is not clear yet where the call goes. We could hardcode something like "mailbox and no external mailbox", but then the question is what about other IVR? Auto attendant and calling card are out because they trigger external calls, do does paging. Conference room is also out, as it is definitevely considered a "rich" application. Service flags? IVR nodes can also redirect calls to the outside world. Star codes? That is a messy topic, e.g. *00123 does call out while *78 does not.