Vodia PBX

There is a lot of confusion about the term "dial plan". In the snom ONE world, it has a limited scope and it deals with the question on which trunk an outbound call that will not terminate internally will be routed. Inbound calls are not routed through a dial plan (at least as of now). There is the simple settings "send call to extension" on the trunk that determines where the call will be sent. Normally, trunks send calls always to internal resources. Trunk-to-trunk calls are the exception, and when you want this you must tell the PBX which dial plan to use and which extensions will be charged (even if there is no real money in the game). For that there is the setting "Assume that call comes from user", which implies that the extension's dial plan will be used to route the call to the outside world. The ERE from above must be put into the trunk setting "Send call to extension" (where extension can actually be anything, e.g. also the external number).

What version is this? Just tested that with 5.0.3 and there it seems to work as it should. I assume that the trunks directory does not show that trunk any more? Is that a global trunk?

You need to import that certificate below as trusted root ca for server authentication. However by default the "Class 3 Public Primary Certification Authority" should be already in your list of trusted root CA if you did a fresh install. -----BEGIN CERTIFICATE----- MIICPDCCAaUCEDyRMcsf9tAbDpq40ES/Er4wDQYJKoZIhvcNAQEFBQAwXzELMAkG A1UEBhMCVVMxFzAVBgNVBAoTDlZlcmlTaWduLCBJbmMuMTcwNQYDVQQLEy5DbGFz cyAzIFB1YmxpYyBQcmltYXJ5IENlcnRpZmljYXRpb24gQXV0aG9yaXR5MB4XDTk2 MDEyOTAwMDAwMFoXDTI4MDgwMjIzNTk1OVowXzELMAkGA1UEBhMCVVMxFzAVBgNV BAoTDlZlcmlTaWduLCBJbmMuMTcwNQYDVQQLEy5DbGFzcyAzIFB1YmxpYyBQcmlt YXJ5IENlcnRpZmljYXRpb24gQXV0aG9yaXR5MIGfMA0GCSqGSIb3DQEBAQUAA4GN ADCBiQKBgQDJXFme8huKARS0EN8EQNvjV69qRUCPhAwL0TPZ2RHP7gJYHyX3KqhE BarsAx94f56TuZoAqiN91qyFomNFx3InzPRMxnVx0jnvT0Lwdd8KkMaOIG+YD/is I19wKTakyYbnsZogy1Olhec9vn2a/iRFM9x2Fe0PonFkTGUugWhFpwIDAQABMA0G CSqGSIb3DQEBBQUAA4GBABByUqkFFBkyCEHwxWsKzH4PIRnN5GfcX6kb5sroc50i 2JhucwNhkcV8sEVAbkSdjbCxlnRhLQ2pRdKkkirWmnWXbj9T/UWZYB2oK0z5XqcJ 2HUw19JlYD1n1khVdWk/kfVIC0dpImmClr7JyDiGSnoscxlIaU5rfGW/D/xwzoiQ -----END CERTIFICATE-----

I would assume that a ERE like !\+4412345([0-9]*)!\1!123! should route all incoming calls to the number +4412345xxx to the extension xxx or to 123 if that number does not exists. Routing calls through the system is a separate topic. For that you need to assume that the call came from an existing extension, so that the PBX knows what extension to charge for the call and what dial plan to use.

We had the same problem with out test installation. There is documentation available for CentOS firewall, e.g. http://wiki.centos.org/HowTos/Network/IPTables iptables is very powerful, but could be easier to use... You can also use netstat to check which port the pbxctrl process has opened. RTP ports are opened on demand, so make sure that you also list them there.

The installers were on the snom web site where they have been removed. We don't support new installations on version 4 any more, only existing version 4 installations can be upgraded to 4.5.1. If you want to make a new installation, you must use version 5, which is available at http://wiki.snomone.com/index.php?title=Release_5.0.

First of all, you can always fall back to extended regular expressions, where you can use \+ to match the plus symbol. However, the real answer is more complex. The PBX actually always tries to keep phone numbers internally in a +-format (this explains why having dial-out prefixes are generally a bad idea). Then when feeding that number into the dial plan, it consults the country code of the domain. For NAPNA (10-digit US and Canada), it presents the number either as ten digits or 011xxx; for other areas is presents the number in 00xxyyzz, 0yyzz or zz-Format (how a naive user would dial the number). That said, there is no need to use + when you have a country code set and you dont have to worry how the user dials the number: Both 00493012345678 and 03012345678 will presented as 03012345678 to the dial plan if the country code is 49 and the area code is not 30. You need to cover this route only one time in the dial plan.

We will address the topic of the hosted PBX in the new year.

Also in snom_710_phone.xml, snom_720_phone.xml, snom_760_phone.xml, snom_820_phone.xml. We'll change the template in the next build. Thanks for the update!

snom ONE is a PBX. There are possibilities like DISA and cell phone twinning where inbound calls on a trunk end up in a outbound call on a trunk. However the product does not offer pure SBC functionality; for this you should check out ACME packet & Co.

The address that the phone delivers.

Obviously a bug sneaked into the phone firmware. Hopefully the next version that snom releases will fix that.

Even the upgrade from Windows 7 to Windows 8 cost money . Or for example, Adobe upgrade cost can be hefty, especially if you have twenty licenses in the company. The problem is that the snom ONE V4 editions were heavily subsidized: selling a 250 extensions PBX for a symbolic price was just too good to be true; V5 needed to correct that. Just for the record: pbxnsip prices were on average significantly higher than the V5 prices.

The first feature sends a HTTP request out when the extension status (e.g. DND, redirection, registration) changes. The URL can be set in the sysadmin setting "Extension Status URL". The format is similar to what you get in the extension status email, just formatted as JSON, When the PBX sends a call to the cell phone, yes then Lync will see that account as busy as long as the user is on the cell phone. You don't need a smart phone for that, even 20 year old cell phones will work with this.

Did you try "Failover Behavior: Always, except for busy tone"?

For the SoHo and mini, you can get a new license key. That should actually work automatically for the minis. Because the SoHo did not have an activation code, you need to contact support to get a new license key. If there is a hardware problem, AFAIK you need to contact the one where you bought it.

Not being the biggest Debian expert, I would think that you have set a default gateway on your static IP, which is probably not a good idea. http://wiki.debian.org/NetworkConfiguration I would have the dhclient assign the default gateway, which seems to be the default. Next time, please use route -n; this is easier to read...

snom ONE does usually not deal with the tones that are coming from the RTP stream. If the other side plays a busy signal, then the PBX relays that tone to the other side. I believe that is okay and actually not the problem here. But I suspect that there is something wrong with the setup of the SPA-2102. I would make a test dial plan that always dials through the SPA and see if that works properly.

What is the output of "route"? Also, what Linux distribution are you using?

It was also on another post: root@pbx:/usr/local/snomONE # wget http://www.snomone.com/downloads/snomONE/mini/libstdc++.so.6.0.13 root@pbx:/usr/local/snomONE # mv libstdc++.so.6.0.13 /usr/lib root@pbx:/usr/local/snomONE # rm /usr/lib/libstdc++.so.6 root@pbx:/usr/local/snomONE # ln -s libstdc++.so.6.0.13 /usr/lib/libstdc++.so.6 The SoHo are on an older operating system; the mini have that newer library already.

There was a problem on the license server if you put a "free" license on top of another license on the same day--the free made it. That should be fixed now.

No "Cell phone". That should do all the magic if you have a snom phone.

Ehh in theory it should be only on one key. Blinking when the call can be retrieved from the cell phone and steady when it can be pushed to the cell.

First of all, IMHO it makes sense to tell the caller that the call is not terminated at the original destination. But if you want, you can of course always replace the WAV file with a short silence or comfort noise (which would affect all users).

Try again...