Vodia PBX
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Posts posted by Vodia PBX
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I really have to get this fixed for our users, who have been pretty patient with me thus far with beta builds and the phone system migration in general. I'll do whatever I can -- my hunch is that it's something internal that I can't fix with a setting, but I would be glad to be wrong.
Sorry, can we open a new topic on this? I lost the context here... What was the problem???
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If you are able to process SOAP requests, it should be relatively easy. Pay attention to the duration element - if it is empty the calls was not connected. Otherwise it contains the numbe rof seconds. More information can be found on http://wiki.pbxnsip.com/index.php/Processi...DR_from_the_PBX.
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I guess you must manually clear the web browser cache then.
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Did you try F5 (reload)?
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I would suggest that you first move all old html files somewhere else. The 2.1 web interface has a lot of updates and you are probably facing version conflicts.
Then what do you need in the html directory?
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As for your first question, do you mean on the software itself or the actual windows domain name?
It is a common trap to rename the default domain name to something else... "localhost" has a special meaning to the PBX - it matches any name. If the name is changed and the PBX receives a request that does not 100 % match, the PBX will send a 404 code. But it seems that was not your problem.
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The best way to deal with home users is IMHO hot desking (http://wiki.pbxnsip.com/index.php/Hot_Desking).
This has the advantage that you are available under one number (also as agent or in a hunt group), but only the phone at home or in hte office rings. Especially for the home phone this is a feature - give your family a break!
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This problem might be related to the transport layer. UDP packets get fragmented when they get too big (around 1500 bytes). I remember switching to TCP or TLS solved the problem.
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In 2.1 you can also change the codecs from the web interface in the system admin/ports section. No restart neccessary any more!
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Well, the PBX does not change the volume. But it might be a problem that the gain on the gateway is too low and users compensate that with a higher volume.
There is a way to calibrate the audio levels on the PBX, see http://wiki.pbxnsip.com/index.php/Gain_Adjustment.
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I think this problem arises when the PBX for whatever reason lost the call and the phone sends a Re-INVITE. In SIP, a Re-INVITE and a INVITE look the same, but the problem is that the Request-URI actually is the phone's own number. That means for the PBX the phone calls its own number, which is the voicemail.
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Is "localhost" your domain name on the PBX? What did you put into "Send call to extension" in the trunk?
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Hmm. Maybe that problem is related to the reason for the transfer provided in the Divert header? What does the PBX say in the SIP packet sent to the Exchange?
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Okay, we did a simulation without the a= header on our phone here and maybe the following version fixes that problem:
http://www.pbxnsip.com/download/pbxctrl-2.1.0.2117.exe
We found something that would explain why the 0.0.0.0 method would not work. Please verify.
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That workaround might actually really be a good workaround - the PBX "dials" the night account number, while on other occasions it just internally switches accounts.
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I think we need a Wireshark trace here...
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We heared about cases where heavy BLF load caused a lot of jitter, but that should not stop RTP!
The PBX writes a log message when it stops RTP dues to one-way audio. The 2.1 contains a new settins "timeout_hold", maybe you can try to set it to a value like 3600 (seconds) and see if the behavior changes.
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Is there a function on the polycom phones or pbxnsip to sound a notice after a call has been put on hold for a period of time? We had an agent put a call on hold this morning and forgot about it.
When using hold, the PBX does not send a reminder (AFAIK there is no way to do this in SIP). So it is the job of the phone. I know some other support hold reminder, but am not sure about Polycom.
FYI the PBX has a park reminder in 2.1. In this case the call is technically disconnected from the phone and the PBX starts a new call to the extension that parked the call.
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No, there is a setting called "List of extensions to watch" (see http://wiki.pbxnsip.com/index.php/Prepare_..._Plug_and_Play).
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I agree that a (possibly temporarily lack) of a license should not mean that configuration data gets lost. And I believe that this should not happen. There is a function that disables the allocation of new extensions, but existing rows are never being deleted.
WHat does happen is that after a restart of the service the PBX is looking for orphan table entries. In that case, it does delete data. But IMHO that should be okay, because otherwise the database could really get messy.
BTW changing the license key does not require a restart of the system.
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That should be fixed in the 2116 build. Try http://www.pbxnsip.com/download/pbxctrl-2.1.0.2116.exe.
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Well, the redirection to Exchange essentially runs through the dial plan. That means you should not dial the mailbox prefix, but the prefix that you are using in the dial plan to redirect calls to Exchange.
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Not yet. We need to talk to Polycom, they would have to add this feature to the firmware release.
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911
in Trunk Setup
I would recommend to use RFC3325 (P-Preferred-Identity) in any case. All the other "ways" are workarounds.
Incoming calls get a busy signal
in Trunk Setup
Posted
For the SIP phones I stongly recommend to use the outbound proxy - then the domain name does not really matter.