Vodia PBX
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it's a mini call centre, at first when customer help the IVR, customer can select 1 to talk to operator, 2 to key in the Identity number. if customer key in the identity number, these will forward to a web database server for veirfying, i managed to set it up via SOAP, however, I am not sure how to setup to let the web server return a message which can be understand by pbxnsip ? apprciate your help please, thanks
Did you already check out http://wiki.pbxnsip.com/index.php/Linking_..._to_an_IVR_Node? Did it help?
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Is this a common prob?
No. The service must (and does) run without reboot. It is a PBX service, rebooting it every day is not an option.
If you need to restart the service because of one-way audio the service is still there, but cannot open media connection. This might be because there are no more ports available (check your RTP port range). The reason can be a lack of resources like ports, but it can also be a firewall that has (for example) too many TCP connections that need to be closed.
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From Auto Attendant to Hunt Group and from Hunt Group to Extension.
I have entered Hunt Group 'From-Header' to 'Group-Name (Calling Party)' otherwise the user don't know how to pick up the call.
When the call is '9xx TEST (Unknown)' the caller doesn't have to enter his/her name before it arrives to the extension. This is what I want to accomplish. That the Unknown callers need to say their name before tranferring to extension.
Having this feature on a hunt group is more difficult than for a single call. It simply does not exist. So far we never thought about it... Usually the hunt group is used for a central number and there (so far) it is not common to ask for the name. But you are right, as more and more automatic machines are war-dialling cheap lines these days, it is becoming a more important feature.
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Anyone else had problems with Snom TLS and SRTP?
Oh yea this topic is a real pain. Whoever invented SRTP did not think about concersations that last longer than 22 minutes - and that calls are possibly on hold for more than 22 minutes.
In 3.0 we introduce a new way of guessing the rollover counter. snom's 7.1.33 also has some improvements with SRTP. We all cross fingers that this will be the last time we talk about this topic.
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Old licenses will work without any troubles?
I believe so...
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What we need is the extension to show the FXO PSTN No (1234567) is this possible?
Do you see the information that you are looking for in the To-Header? Check the SIP INVITE that goes to your phone. Maybe it includes already the "To" header. Then the next step it to think about getting that into the From-Header...
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It does not make a difference - from what I can test - it's transparent.
The problem with the address book matching is that in many cases when the call comes in it is not clear where it exactly goes (at least at the time when the address book query happens). For example, if the call goes to the auto attendant then at that time it is impossible to use the user address book yet. But your case sounds more easy.
How does the call come in? Sounds like there is a problem finding the right domain.
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One of our customer wants to use a client with GSM codec. Does PBXnSIP support GSM codec?
Yes. To be more precise, the GSM 06.10 FR codec (there are also others). The codec number is "3" or just "gsm".
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I' currently using the pbxnsip version
Version: 2.1.10.2474 (Win32)
I noticed my server will experience one way audio perpetually every monring.... and requires a restart of server/services before getting it to work normally.
Whow. Restart of the server or just the service? If you also need to restart the server then check if there is some virus or malicious program consuming a lot of resources (memory, handles, ...).
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Site says: Extensions: 10/10 Accounts: 19/20
however i only have 9 extensions and a PAC extension.. i thought the PAC did not count against license?
What is a PAC extension... ? Extension is extension...
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Does anyone have some experience with the provisioning function within pbxnsip (version 2.1.11.2484) and snom phones?
I have created a test environment for approving before I replace all the phones in my office.
Currently I am testing with Snom 300 phones and I am not able to get the buttons provisioned thru pbxnsip.
Can anyone help me get this working?
I have also placed the file snom_300_fkeys.xml into the html directory but that also did not change anything.
(http://forum.pbxnsip.com/index.php?showtopic=1002&hl=button+snom)
The 2.1 branch does not support the provisioning of the buttons of the 300 phones. Maybe it is time to "officiallly" start the 3.0 beta runs, because 3.0 fixes these little known problems.
Alternatively, you can set up your own fkey provisioning file. Take a look at the generated directory, and then you can just move these files into the tftp directory and edit them there locally.
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How can I switch-off e-mail notification 'Recording available' to admin e-mail. PBX sends report about every new recording, which I don't want to have to.
You can't... Workaround if you don't like the emails is to set up a email rule to send them to the trash can.
I guess we need to introduce a setting that says what emails should be send which emails not.
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Is there any official download link?
No, not yet... You will have to live with a beta image. Send me a PM if you want a link (and what OS).
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I thought that way, but I can't find such parameter in agent group settings...
You are right... That feature exists only in 3.0. If it is really important to you, then you should consider upgrading.
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Can I setup agent group such as it'll play ringback tone instead of MoH, when the user is at the queue and no free operators available? I need exactly agent group, not hunt group.
Just set up a MoH that uses the "ringback.wav" as MoH file. Then you can assign that "music" to the agent group. The PBX will think it is music, but the caller will think it is ringback tone.
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Does the extension to extension intercom feature work on Polycom phones or SNOM only?
It also works on the Polycom - however you need to provision then through the PBX. Otherwise the setup of the configuration files will be really difficult.
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Till now I don't get it manage to working. I entered the SAME (see attached picture) information I entered in X-lite in the SNOM M3 and it's not working. The server is at the moment with a direct connection and no firewall, to be sure it's not a firewall issue.
What is extra necessary in the settings for the M3 to have an external extension working?
The only thing that comes to my mind is that the server is "local", because the PBX does not want that the SIP phone tries to get a routable IP address (through STUN or other more-or-less buggy methods).
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It shows that the traffic times out, it goes nowhere. I dial the number and sweet silence of the lambs. The strange thing is that inbound calls are fine. The VPN with the pbx can make inbound and outbound calls, just not the Snom 370 on its own using VPN, very strange indeed.
Are you using outbound proxy? I recommend to always set the outbound proxy, unless you really want to call SIP URI in peer to peer mode.
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I'm using 2.1.1.2484 on Debian 4.0. There is error forum my log file:
[0] 20080628190134: load: Table acd not found.
What's this mean and what should I do?
That is a problem with one of the web interface files. Not very serious. It means that the web server tried to load something from a table that does not exist (is was acds). We need to update the file and fix that in 2.1.12.
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I am aware of the registry changes that should be made in order to set DSCP on outgoing packets, however microsoft also installs a QOS packet scheduler by default, and binds it to the NIC. Is this good, bad, or indiferent from the PBXnSIP point of view? Just wondering if I should leave it, or not.
IMHO the packet scheduler makes only sense if the traffic that leaves the computer can exceed the speed limit of the NIC. For example if the NIC is connected to a T1 and you run both voice and data on it, well then a packet scheduler is really useful. If you have a 100 MBit NIC to the LAN and the computer is not a busy file server well then there it does not make sense to me.
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I was wondering if there was a folder I could put files into so that they would be accessible by all phones via HTTP. I am trying to make images available to phones. Currently I need to install IIS on an alternate port to do so. I need to leave PBXnSIP running on 80 for the few snom phones on my network since they will not get their info via https. I have tried putting my files in the html folder, however if I am not already logged in, it redirects to the login screen, and does not give the graphic.
You can put the files into the tftp directory. Then you can access them with http://<ip-adr>/tftp/filename.
No need to run IIS for that.
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That is precisly the problem. We have to have tel:alias in order for incomming calls to reach proper domain, remember we are running in hosted mode with many domains. On top of that our povider requires that on outbound calls we send 10 digits to them. Only then they pass it along, if it is not 10 digit number they send the number asociated with PRI (our office main number). If we use 10 digit tel:alias our gateway doesn't know how to send incoming calls correctly since it is a local 7 digit call. So now everyone's caller ID appears as our main number, so we get a lot of calls from people that are "returnig" their calls since it appears that it came from us
Remember that you can also use two (or more) tel:alias in one account. The first one will be the one that is used as ANI, but the second one is also used for inbound matching. Maybe that helps to solve the problem.
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I have the 2.1.11.2484 (Win32) I guess is the last one and I still have to dial 1 confirmation
Ehh.... Okay, that version does not have this. 2.1.12 will have it
. Then you can add the parameter "connect=true" and it will start dialling when the handset it lifted up. -
Is this by design?
Nope. Does it make a difference if the contact type is white or black or just transparent?
. Then you can add the parameter "connect=true" and it will start dialling when the handset it lifted up.
problems with Microsoft speech server
in Microsoft Speech Server
Posted
Hmm. I know that a few folks got the Speech Server working, however I beliebe that the 2004 version might be a little bit outdate (but I am an the big Microsoft expert here).