Vodia PBX

Q2: Well, how? The image is just a image, difficult to link pixel to links. But I get the point. The PBX also sends out CDR with the MOS attached, you can (e.g. in Outlook) set rules when the PBX sets the "alarm" flag for a MOS score. Q3: Absolutely. Every CDR email has an attachment with a MOS overview. Q4: AFAIR yes that is possible in the system settings. Q5: Again, the details are in the CDR emails. Q6: You mean ITSP? I have not seen one with MOS scores, espeically those who don't own the copper or fiber to the customer...

In Linux, critical events are also sent to the syslog. In Windows thats not trivial and we dont have it yet. Apart from that, in a productive system, you should always lower the log levels which wil show you only the most relevant log messages.

Unter http://www.pbxnsip.com/download/pbxtsp11.exe sollte das TSP noch verfügbar sein.

Hmm. I assume you are using the latest build (Epsilon)? Do you have your server running on an address where we can try to connect and send out an email from a debug system? We definitevely want to close the chapter.

I suspect the problem lies in the new cell phone handling logic. When an extension is connected to a cell phone, the PBX does not hang up and instead plays a voice menu what other actions are now possible on the cell phone (which is a feature). I am sure that the PBX thinks that the call is connected to a cell phone, as it "assumes that the call comes from user so-and-so", which would perfectly explain it. We have to check if we can do the workaround that we clear the extension on the leg after the call was sent to the trunk or we have to introduce another internal flag which says "hang up or play ivr".

Looks like the problem is in the MKI area (master key index): a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pmP4qAsGAw1ojK/EYz6yU076pauMb79BPc95T+iw|2^31|3:1. The PBX simply does not support this at this time. Can you check if there is a workaround possible on the gateway? If not, we'll have to add this to the PBX, but this will take some time and some testing.

How is your pnp.xml file looking like from the web interface from the PBX (admin/web page control/phone pnp/pnp.xml)?

Hmm. I think we need some SIP traces from the phone. First, does the phone send anything out when you press a button? And lets say a pickup becomes possible, does the PBX send a MESSAGE to the phone?

Das Ding hat schon ein paar Jahre auf dem Buckel, aber es gibt ein TAPI service provider für snom ONE: http://kiwi.pbxnsip....ervice_Provider. Ich bin mir nicht mal mehr sicher ob das überhaupt noch funktioniert und es überhaupt TAPI noch gibt, aber damit kan man aus dem Outlook heraus Nummern wählen.

Seems like we are getting closer... Does the gateway play back the stream from the PBX (one-way secure audio)? Maybe the problem is that the gateway has a dfferent opinion about the rollover counter...

What are these keys programmed to? We had (actually years ago) problems when the shared lines got stuck and would not come back after a minute or so. A reboot of the phone should actually solve the problem as well, as it should re-program all buttons again. Are you using PnP for the phone?

Please send the activation code to us (e.g. in a private message on this forum) and we can figure out if the key has been already activated. There was also a problem with the upgrade from snom ONE free to blue/yellow, which should have been fixed on the server side in the meantime. The other thing is that the older versions (before 4.5) had problems with the output generated from the server; that should be fixed with the new 4.5 version.

Thats probably also a problem with the SDP negotiation of the SRTP key material.

Sorry this forum is about the snom ONE (which is a IP-PBX from snom). For general questions about snom 370, you should post it at forum.snom.com.

The M1000 proposes several m-lines in the SDP; the PBX happily picks the first one which does not not have SRTP. If you want that the PBX does SRTP, the first m-line should have the crypto attribute in it (AVP or SAVP does not really matter to the PBX).

Technically, we are able to generate a key for that. But only the sales team can answer that question...

Ich bin der Meinung dass man nicht alles aus der guten alten PBX-Zeit übernehmen muss. Diese Präfixe haben ein riesiges Durcheinander angerichtet, weil nie weiss wieviele Nullen man wählen muss bevor man endlich raustelefonieren darf und so mancher ist statt in Reutlingen irgendwo in Tokio gelandet. Hintergrund was dass die Null "das Amt holt" (Shared line appearance), was bei SIP bestenfalls simuliert wird: Es wird definitiv damit keine Leitung belegt. Auch ist es so, dass inzwischen jeder ein Handy hat und dort kann man sich auch kaum vorstellen, vorher eine 0 zu wählen und es hat sich auch inzwischen jeder daran gewöhnt, die grüne Taste zu drücken damit es losgeht. Ausserdem muss man sagen dass es Overlap Dialling (Amt meldet es wenn die Nummer vollständig ist) bei SIP in der Praxis nicht gibt und selbst im Labor nur endlos Probleme auftauchen. Und zum Schluss gibt es noch das Problem wenn man einen Anruf verpasst hat und zurückrufen will: Amt holen oder nicht? Man kann snom ONE dennoch so einstellen (verstellen) dass die 0 schlichtweg ignoriert wird wenn man raustelefoniert. Das gibt einigen das Gefühl jetzt die Anlage wieder wie früher bedienen zu können. Aber es gibt halt die vielen Probleme mit Rückruf und Adressbucheinträgen, und so würde ich sagen: Die Mitarbeiter sollen einfach die Nummer wählen die sie wählen wollen und danach auf die grüne Taste (oder # bei ATA) drücken. Die Anzahl der Tastendrücke bleibt gleich.

Vorher backup machen... Man weiss ja nie!

Sollte man eigentlich auch über das Webinterface auswählen können:

Did you plug and play the snom 300 or did you set the settings on your own?

What version is this? We have "Intercom" in the button dropdown (running the latest version)...

So you put the call on hold on the snom 300? Do you have audio_moh installed (I guess so if this is a snom ONE plus).

Ja das ist ein bekanntest Problem; die Lösung ist etwas tricky da die PBX From/To umdrehen muss wenn der Call zustandekommt. Es gibt ein offenes Ticket zu dem Thema, es wird nicht vergessen.

If you upgrade from a previous version, you might possibly keep the old snom ONE free license key. Either try a fresh installation or reset the key and then re-apply the snom ONE free key; then you should be able to register third party devices.

Well just send it to sales (at snom dot com). Of course if you just want to try out snom ONE, you can as well take the snom ONE free. But if you are unsure how the installation scales, 3rd party and so on, then it makes sense to use a snom ONE blue with a 30 day expiration.