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Posts posted by Vodia PBX
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Phone and switch MAC tables will cause problems like this, especially if MAC learning is enabled.
You mean the ARP cache does not get updated? That would be a huge surprise and IMHO a bug.
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Would you see any reason why the phone would not find the PBXNSIP server? Maybe there are any settings that can't simply be transferred from one machine to another.
Probably the outbound proxy is not correct any more. My suggestion is to use the same IP address on the new machine as the old machine, this way you can keep the phones untouched.
An alternative would be having a completely automatic plug and play. Then you may have to change option 66 on the DHCP server, and that's it.
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Before we upgraded to the latest version, I had a$ xxxxxxxxxx in the Explicit Remote-Party-ID. Again the x's representing our actual phone number.
Maybe just put the xxxxxxxxxx there.
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We are looking for a way to forward all our technical support calls to a cell phone when needed. The calls come through our main AA and into another queue that the techs login to. We tried having a user login to the queue then forward all the calls from his ext to the cell that didn't work it only forwarded calls that came directly to his ext. It would be great if this could be done from the web page also so if a user is going to work from home he can setup the ACD Calls all to go to his cell phone.
You can redirect the calls from the ACD to an external number, that should be no problem. If you redirect from a ACD also to an extension, then the cell phone redirection rules apply again and the call will fork to the cell phone as well.
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I tried to leave the Trunk DID blank, but the caller id when receiving a call from my voicemail came up as unknown again. So is there something I can do about this problem?
Do you remember what you had in that field before? ...
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Sounds great! Anything that you would like to change on http://wiki.pbxnsip.com/index.php/Polycom to make other's life easier?
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We got rid of these "$f $i $x $y $z" strings in the trunk. I think the new name is now "Trunk DID". Normally, it is okay to just leave it empty, the PBX then will automatically fill something useful in.
It is debatable if that was a smart move. On the one hand, it caused a lot of support and problems with too much programmablility for nothing. On the other hand we have an upgrade issue, and now we need to explain everyone .
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Whow. Indeed, that default certificate expired almost two years ago...
If you want to get rid of these errors, you better get your own certifate anyway. There is some info on http://wiki.pbxnsip.com/index.php/Getting_...lid_Certificate.
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That means that the PBX received a new media stream on the RTP port. It is allowed according to the IETF, but raises questions. A legal case could be that the other side uses an external music on hold server and sends music on hold out of the original media context.
"Illegal" cases would be that there are really several audio streams running on the same RTP port. Do you have enough RTP ports open on the firewall? Or maybe there is a device that just did not stop sending audio streams?
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http://www.pbxnsip.com/protect/update-2895.tgz has some more flags, e.g. for the CPC duration.
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You don't have to use all of the fields, some of them are only of "internal" interest.
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When a call redirects to my cell phone, it shows the original caller ID which is good.
Congrats! You are a lucky guy. Most of the PSTN services do not allow this. What PSTN termination are you using?
Is there any way to set a prefix to the caller ID, so I know the call is coming from the pbx rather than someone dialing my cell directly.No, at the moment not.....
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Is the log call is in the folder "cdr"?
Well if you have to, you can use the files in that directory to generate your CDR. What OS are you using? Where is the mySQL? Are you using something link LAMP?
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The trunk settings have changed slightly. The caller-ID presentation has changed, this can be seen on http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk. I guess that is the problem here.
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So you mean after manually entering the settings it does not work? Maybe we need to step back to look at REGISTER requests?
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I have a Siemens Gigaset sx255 and i want to connect it to my voipbox. Is that possible?
I think so - maybe you should start a new topic on this.
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You may have to re-check your trunk settings: See http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk.
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Did you set the tftp server to the address of the PBX? Also, do you have any extension set up for this MAC address or do you use a * to match any address?
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The good news is that we have at least the option to change things... Not that we like it, but being able to is a good thing.
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If you don't habe a routable IP address, you will have a lot of pain. SIP is not HTTP.
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If you can try the update image at http://www.pbxnsip.com/protect/update-2894.tgz. But make a backup of your old config, maybe even of the whole file system.
This image has a new option for "CPC Duration" - try a higher value than 80 ms. Hopefully that helps sorting out this problem. You might also want to turn off Polarity Change detection and Busy tone detection.
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I tried using both of the dial plans on my phones but I am still getting the message that the feature isn't available.
If you hear the IVR annoucement "This feature is not available" then the phone obviously connects to the PBX. Then it is not the phone's dial plan, in this case you have to continue the search on the PBX.
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The problem is most likely the dial plan in the phones. There is some stuff on http://wiki.pbxnsip.com/index.php/Linksys, maybe try changing the dial plan on the phone and see if that helps.
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Start: Last Sunday in March at 2 am local time
End: Last Sunday in October at 2 am local time
Sounds like central european time, just two hours ahead? If that is so, it should be a piece of cake.
IVR Outdialing
in IVR Node Setup
Posted
The setting for the dial plan in indeed missing for the IVR node. The IVR node can only call internal numbers that do not require a dial plan.
Feature will be available in the next version.