Vodia PBX

Well there is a roadmap and there is a idea map... For the latter it is good to chat about what could be done

When selecting fixed length CMC codes, the PBX does not check the address book. It just takes the codes and stores them. But thats probably not the point here. I guess the problem is that the call is not connected yet and the device that you are using does not send DTMF during the "early media" phase.

Right now thats not possible, you can only listen to calls from extensions. What you can do though is to record all phone calls to the AA, either write it to the file system or have the system call you. Interesting idea though. AA may be one thing, but listening to people talking to my/other mailboxes and to ACD sound a lot more interesting. Permission concept is the key here!

Just tested it here... worked... If you use the "P" flag make sure that the user has a PIN and 4 digits codes for CMC should also be fine. You must be calling from an extension.

If the IVR does not stop playing the message when you enter something, the problem seems to be more fundamental, you might have a problem wíth the DTMF detection. Does DTMF work when you for example call the auto attendant?

Hmmm. Yea it seems the pnp file was wrong, it should like that: <file name="snom_pa.xml" encoding="xml"> <pattern>snomPA1.htm</pattern> <vendor>snom</vendor> <anonymous>true</anonymous> <protocol>tftp,http,https</protocol> <pnp-vendor>snom</pnp-vendor> <pnp-model>snompa1</pnp-model> <pnp-version>^[78]</pnp-version> <pnp-content-type>application/url</pnp-content-type> <pnp-url>{https-url}/snomPA1.htm</pnp-url> </file>

Well the point is that now the PnP files can be edited directly from the web interface, there is (almost) no need any more to do this in the file system. If you look carefully, you will actually notive that the database stores not just the web page, it also stores the domain and the username. In other words, we are for far away from provisioning different file versions per domain and per user. IMHO that is worth sacrifying some backward compatibility with the custom PnP files.

Sounds like you can play funny games in big offices with that feature

Ehh redirection... So lets say A calls B, and B has set "redirect all calls" to C. Then if C is busy, the PBX will offer a camp on (A does not know that B has redirected the call). IMHO it would be reasonable to say wait until B gets available, not caring about C. When the call gets redirected again, well the PBX needs to do that otherwise we could create a loophole for calling someone direct. If we exclude camp on when the call is redirected, this is also not really solving the problem; B could set the redirection after the callback was registered.

I would check the log file of the phone to see if the file gets downloaded and for example the format is right.

http://forum.pbxnsip.com/index.php?app=forums&module=post&section=post&do=reply_post&f=14&t=4090&qpid=18723 deals with the same problem, but for snom 821.

The whole topic of sending out multiple messages is messy. We'll clean it up in the next major release. Right now you have to live with the limitation that if you want to delete the message after sending it out, you can have only on email recipient. If you want to have more than one, you'll have to use the workaround with an email list, where the server takes care about sending it to more than one recipient.

No it is not the keypad activity. It is call-based, not presence-based. This makes it possible to see if someone becomes available for example on his cell phone. Well of course it should work. I think many people find it annoying and turn it off, that's why probably the exposure is lower than it could be.

Okay, at least we know that DNS is the problem here. If they change the IP address, you'll know pretty fast... Hopefully you remember that the neccessary change is the IP address. Of couse it would be interesting to know why the DNS does not properly. IMHO it should not be a problem of the PBX, probably some problems with the interop of the next DNS relay server. We have to keep this in mind and if other people report similar problems, we have to see if we can change anything on the product side...

You probably provisioned a "dial plan" on the phone (notice the name clash with the dial plan on the PBX). That tells the phone to accept either 1xxxxxxxxxx or [2-8]x as complete numbers. The PBX provisions this dial plan when you select this in the domain PnP settings. The alternative to this is to let the user hit the enter key instead, and then they can dial without the leading 1. The number of key presses is the same and this is like the dialling process on a cell phone. I would prefer this kind of the dialling, but some customers prefer the 1xxx... way.

In general, extensions should have between 1 and 5 digits. When translating numbers, the PBX translates numbers with at least 6 digits into the global format, but leaves numbers with max 5 digits untouched. Before running calls through the dial plan, the PBX checks if there is a account with exactly the same number. Only if there is no match (and there is no exception regarding emergency numbers) the call runs through teh dial plan.

Bei der neuen Version (http://wiki.snomone.com/index.php?title=Upgrades) 4.2.0.3981 sollte das automatisch gehen. Die Datei snom_821.htm würde ich wieder löschen oder umbenennen; sie ist bereits im neuen Build enthalten. Beim 821 einfach die IP-Adresse der PBX eintippen bei "Advances/Update/Setting URL", neu starten und danach sollte es alles von Geisterhand automatisch gesteuert werden.

The problem is that the phone insists to use G.729, but the snom free does not come with that codec. As the name suggest, snom ONE is for free, but the codec is not so we had to take this out. If you need G.729, you have to choose snom ONE yellow, blue or green. Alternatively, you can use GSM or G.726 codec, which also compress the voice almost like G.729.

The PBX loads it again after playing it once (the WAV file itself does not loop). I would keep it to the original size, which is a good tradeoff between loading and memory size.

Okay, so what is wrong? The duration statement in the email? Or the message got cut off prematurely? What is the size of the file in the file system? 15 seconds would be 15 * 13200 / (8 * 1024) = around 24 KB. It seems that the Linux version and the Windows version have different GSM headers set due to their operating system differences, but it should not randomly make the files shorter.

Let me try to repeat. You received the voicemail in am email attachment and there it was 53 seconds, while on the file system it was only 15 seconds? Was the content the same? Was there something in the long voicemail after 15 seconds (no silence)?

Maybe there is a misunderstanding. Trunks can register, but not receive registrations. Extensions receive registrations, but not trunks. You dont have to register the device, just send the traffic there any you are all set. If you want to send calls to the PSTN through a gateway that is e.g. behind NAT, then you can register the device to an extension and in the dial plan you can use the option to send a call to an extension.

Es gibt zu diesem Thema schon Erfahrung: http://forum.pbxnsip.com/index.php?/topic/3774-admin-console-setup-for-mac-os-x-server/. IMHO ist es okay dass die PBX nur dann den Betrieb aufnimmt wenn auch Port frei ist. Bei LDAP kann man geteilter Meinung sein... Anyway, ich würde mal versuchen die PBX manuell zu starten und zu sehen was auf der Console ausgegeben wird. Notfalls einfach mal folgenden Text in pbx.xml kopieren (im gleichen Directory wo auch die PBX läuft): <?xml version="1.0" encoding="utf-8"?> <pbx-config> <ip_http_port>8080</ip_http_port> <ip_https_port>8081</ip_https_port> <ip_ldap_port>8082</ip_ldap_port> </pbx-config>

Ich würde folgendes machen: Den Web-Server mal kurzzeitig runterfahren, die PBX starten und dann im Webinterface der PBX einen anderen Port einstellen. Dann PBX neu starten und den Webserver wieder rauffahren. Alternativ kann die PBX manuell mit --http-port 8080 --https-port 8081 --log 9 gestartet werden. Dann gibt es auch Ausgabe auf der Konsole, falls noch andere Ports im Wege stehen. Der nächste MAC Installer sollte das Problem beheben oder zumindest deutlich vereinfachen.

Is it no audio or one way audio? If you have a setup with multiple IP addresses (private, public, VPN, ...) the routing could be a problem. First step should be to get it working in the LAN, then if that works you can complicate the setup and put the device behind NAT.