Vodia PBX

The one that you are redirecting to is always an exception (typically the secretary). That solves the problems in most cases. Apart from that you can control who is allowed to call you in the permissions tab, maybe that also helps.

The SIP log looks good/beautiful. To me it seems that the carrier is generating the prompt. Maybe there is a "1" missing at the beginning of the number? Try to connect a analog phone to the line, dial the number through the analog line and see if you get the same answer.

Oh the CS410 is already some time ago. The default config for the CS410 had it all right, did you see that when you looked at the config? What SW are you actually running? You can try to send the traffic to an external destination, just to see if the traffic actually leaves the PBX. Maybe this helps to narrow the problem down.

Thats incorrect. snom ONE can deal with 2048 bits. You need to install that as server cert, and you must include the private key as well (everything base64 encoded). if you import a certificate chain, make sure that the Root CA is at the top and the intermediate right below it. You should see that also in the wireshark trace then.

Naja, das ATA186 ist ja schon etwas "betagt". Aber ein ATA sollte aufgrund der beschränkten Funktionalität relativ einfach zum laufen zu bekommen sein. ich würde bei der PBX das SIP-Logging auf die IP-Adresse des ATA einstellen und mir mal die SIP-Pakete anschauen (Registrierung, ein- und ausgehende Anrufe).

Is this a analog line? If the answer is yes, then the NBE has to detect the disconnect tone because in the analog world, there is no signaling for the hangup (unless you have polarity change on the line)

Try to whitelist 10.2.3.4, maybe we are in trouble with the automatic blacklisting here.

Well the other "classic" would be that phones are registered over the Internet and the connection is not reliable. Lets say the PBX sends the INVITE, but the CANCEL does not make it. In that case, the PBX would send hte call to the VM though.

Well, thats the problem when modifying something in the web interface. When the upgrade introduces new texts, those are missing in the customized version. I agree, this is something that we need to look into.

Are they twinning their extensions with the cell phone? Maybe that is causing the issue, the PBX might be waiting for the users to press a button but it is only the cell phone mailbox talking to the PBX.

Probably. I would definitevely make a backup of the PBX directory, so that in case anything gets screwed up, you can move back. Then find a time when there is no traffic and then you can try to change the setting. With changes to the network configuration you have to be a lot more careful as it is not so easy to undo them later. Also, I would try to paint a diagram on how it should look like, so that the changes you do on the routing make sense.

Sieht so aus als NBE das SDP mit dem T.38 nicht mag, obwohl es eigentlich nicht schlecht aussieht. Ich denke wir müssen hier mal den Sangoma-Support einschalten...

Wrong forum? Try forum.snom.com ...

We tried the same thing (with Linux) in our own office. It is a difficult topic. The way to differentiate RTP from other traffic is the DiffSrv bits. You can set them in the PBX up (take a look at the pbx.xml file). The default values should already be okay. For inbound traffic, whoever is sending the traffic needs to know where to send the packets, that must be done by the IP address. The consequence is that you need to assign a specific IP address for the RTP which is different than the IP address for the SIP, HTTP and other protocol ports. There is a setting "Bind to specific IP address" where you can specify to what IP address the PBX should bind the RTP ports (for IPv4 and IPv6, respectively). This should solve the problem that all RTP traffic will be send from and to a specific IP address, which you can connect to the special line for RTP. We tried some time ago with Linux, but were not able to tell the OS to stick to the IP address when sending traffic out. Maybe thats easier in Windows.

No. In Windows you will miss the installation of the service. In Linux and the other distribution you will miss the automatic startup of the daemon. The suggested procedure is to make a fresh install on the target computer, then rename the PBX directory to something like "snomONE-old" and put the directory of the old server on the new server. Then test by doing a reboot if everything made it fine.

4.5 introduces much more flexible headers. While that is generally a great thing and gives you much more flexibility to deal with all kinds of service providers, it comes with the price that such upgrades might not work as smooth as they should. Do you remember which mode worked in version 4.3 for your provider? Then you should try to set the headers the same in 4.5 and see if it works again. It should be automatic during the upgrade, should.

IMHO that is very well spent money. Echo is very annoying and makes a system sound inferior, on both sides of the call. I wonder how they can sell the card without echo canceller!

Okay, wenn das Problem auch bei internen Gesprächen passiert, liegt es sicher nicht an der Firewall oder am Provider. Wir hatten neulich einen Fall, wo jemand der PBX nur 10 RTP Ports spendiert hatte; ich hoffe das ist hier nicht das Problem. Ansonsten wäre es gut, wenn das Problem bwzüglich der Endgeräte weiter eingekreist werden könnte. Z.B. wäre es möglich, sich mal die CDR anzuschauen (im Working Directory der PBX), da gibt es auch Statistiken wieviele Pakete gesendet und empfangen wurden. Vielleicht fällt ja dort ein Muster auf, z.B. bestimmte Firmware-Version und/oder Typ.

Meine Vermutung ist: die Firewall hat keine UDP-Ports mehr frei. Vielleicht gibt es ein Datenblatt wo drinsteht, wieviele Ports die Firewall hat oder sogar ein Log, wo ein Eintrag gemacht wird wenn keine Ports mehr frei sind. Die PBX macht für jeden Call eine Statistik, wieviele Pakete rein- und rausgegangen sind und wohin. Das sollte man im Directory cdrt sehen können, oder wenn für jeden Call eine Email verschickt wird, befindet sich diese Information im Anhang.

Guter Hinweis, darauf wäre ich auch nicht auf Anhieb gekommen.

We just added a license option that allows the PBX executable to behave like a pbxnsip image. That reduces the overhead to generate two images for all platforms.

hmm... that might be the case. I guess we need the snom support to chime in here...

"INVITE Response" heisst dass die PBX das als Antwort zurückbekommen hat. Aus irgendwelchen Gründen scheint die Fritzbox das zurückzuschicken. Es gibt eine alte Anleitung unter http://kiwi.pbxnsip.com/index.php/AVM zum Thema Fr5itzbox, die vielleicht weiter hilft.

Interesting question. I believe according to the RFC, this is the "intended" behavior. I know aout customers who want to to ring for two hours, if neccessary. But it would be good to stop ringing if the TCP connection goes down. The alternative would be as you indicated a settings how long it should ring max. There is a setting ring_duration (see http://wiki.snom.com/Settings/ring_duration) that might do the trick, although it seems it redirects the call (maybe you can redirect the call into nirvana after five minutes).

Sounds like a good idea. Someone will contact you shortly.