Vodia PBX
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I was just wondering what the "last invite =1" thing is?
Well that would mean that the INVITE is already out.
There is a new version that has more logging in this area: http://www.pbxnsip.com/protect/update-2898.tgz. Maybe you have the chance to give that image a shot.
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Thanks for your answer, but how can I do that Pulver get a session border controller?
Well write him an email and see what happens!
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Well the PSTN gateway is really a seperate subsystem. Just like you connect an external PSTN gateway. The communication runs only via SIP.
If a CO-line on the PBX gets stuck, then that means that the gateway did not send a BYE. So if you do logging, make sure that you are watching the SIP traffic on the IP address 127.0.0.1.
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[3] 2008/03/25 21:02:47: PSTN: Channel 1 going to RING
[5] 2008/03/25 21:02:51: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT
[5] 2008/03/25 21:02:51: PSTN: Received on 1 Caller-ID 3256900821
[5] 2008/03/25 21:02:51: PSTN: Received on 1 Name JACKS DEBORAH
[3] 2008/03/25 21:02:52: PSTN: Channel 1 going to NO_RING
What surprises me here is that channel 1 obviously receives the caller-ID while in the "RING" state... Is that possible? I thought the caller-ID is sent in the pause between the first and the second ring?
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You could hijack one of the existing prompts (e.g. "for support press") and use an unspellable pattern as input (e.g. "x"). Just a wild idea.
With "hijack" I mean overwrite. Then select it in the AA as one of the annoucements and as direct destination put a "x" there.
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You could hijack one of the existing prompts (e.g. "for support press") and use an unspellable pattern as input (e.g. "x"). Just a wild idea.
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2.1.6 supports the recording of conferences. You must use a scheduled conference for that. When you create the conference, check the recording flag. Then the conference recording is available for the moderator (the one who created the conference) from the web interface.
Needless to say, those files can get big. The conference must be deleted manually from the web interface to delete the recording.
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STUN works only in < 80 % of the cases, leaving the remaining > 20 % as a support nightmare. We gave up troubleshooting all kinds of one-way audio problems and explicity removed the support for STUN-allocated identities. Our life and the life of our customers became much easier after that. And we saved a lot of money buying all these different DSL and cable routers to find out what the problem was with this and that installation.
Pulver must get a session border controller. That is what practically all SIP service providers are doing.
Or they should start supporting IPv6 addresses (no more NAT the way it was done in IPv4). If you don't have a routable address, don't expect that someone calls you
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Anyone have a clue on this one???
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.
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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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Voice Prompt - call by name
in Auto Attendant Setup
Posted
Ehh... You are right.
There is still one possibility. A auto attendant will immediately do to dial by name if you put "start" into the place where you currently have "411". If you use two auto attendants, then the first one can make the annoucement as described above, then redirect the call to the second attendant (which has an unspellable name like "dialbyname") and bingo - we have the perfect annoucement.