Vodia PBX

Hmm. This is strange. Can you log into the box (SSH) and check what is in /etc/sipfxo.conf? There should be a line with "SIPLOG", and it should be set to "true". If not, just use "vi" to edit that file and give it another reboot...

So you are using the internal mailbox? Or does the call get redirected to an external voicemail system (check the setting "External Voicemail System" in the domain settings). I assume you already checked if the mailbox of the extension was accidentially disabled and hit the save button on that page - just to be sure that it is really saved?

There should be something in the log with "PSTN:" in it. Check the logging page if you have turned PSTN event logging on (you may turn other events off to reduce the number of messages that you see in the log file) and make sure that the log level is high enough; after that you need to reboot the device before it takes effect.

Seems like it is impossible to stay on that page. Seems there are people out there who fear a fair comparison.

Maybe give version 2.02 a shot. Should be available now; apart from that maybe the distance or receiption is a problem. Although it is digital, maybe the device wants to reduce potential noice and lowers the volume. Just a wild guess ...

It should be fixed after 3.3.1. Ask support@pbxnsip.com for a build. Hopefully we can make it GA pretty soon.

Okay, version 4 will have this.

Absolutely. The PAC will be using this as well. Now think about the PAC running on your cell phone. Bingo.

We had a AGC in mind, but giving the conference participant the explicit control could also be good. Though I doubt that Joe Av will ever use it, for him something automatic is easier. The other point that we had in mind was a XML screen where a phone user can see who is in the conference. That would depend on the phone type but it would be supercool if the moderator can see on the screen who is in the room. And the other point was calling people into the conference. The moderator could add a list of numbers that should be called into the conference when the start time has been reached. And we have to think about such simple things like no conference without moderator.

What about inband DTMF? Is that an issue for you? We had a lot of issues recently and added inband to out-of-band transcoding to get that working....

Are you using plug and play? If not, consider provision the phone once with the PnP mechanism and then you can see what dial plan the phone is using. Don't forget to set the PnP preferences in the domain.

Very nice!

Yea, that is "normal". It looks like the gateway stops sending media, mayb because of VAD. I believe it makes sense to look more in this direction.

Not to scare you, but... There are also other ways of DoS. If you are in the LAN, try ping -f, maybe from more than one computer. Okay, that's easy. The other nasty thing is spraying packets on the PBX, just to consume bandwidth. Especially for systems that don't have too much of it (e.g. sitting on a cable modem) and who don't have a QoS mechanism, it is easy to tease those installations badly. The experience is that the audio quality will suffer significantly. You will be chasing this for more than just days. We had a case some time ago, where the other party accidentially crashed the application, but the media part was still alive (of course that other party was not running the pbxnsip PBX ). However, it was serious because that media server was sending RTP well-formatted, constantly on the PBX. It was simply comsuming so much bandwidth that we could not make phone calls over that line without mmm-aaa-jjj-ooo-rrr quality problems. Just before we were ready to call our service provider and beg him to blacklist this IP address, the media server had mercy and rebooted. Even blacklisting that address on the PBX would not have changed anything. Those guys who believe that ENUM and peer to peer will be the future should think about this scenario. Check out MPLS. It is not so stupid.

So far this is not very common. But I am afraid it will be more common. The IP address white and black list were done for a reason. Yes, after finding out it saved your day. What we can learn from this case is that we need to handle this more automatically. IMHO it is not enough to use iptables in Linux to address this problem. What we need in the next version is an automatic blacklisting of addresses that fail to authenticate for so and so many times. Today the biggest limitation is usually the speed on the link to the Internet (for example, the typical CS410 case will be like that). However, as people are moving to hosted environments with very high speed links, that "natural" protection will not be there any more.

I would exclude that is has to do with the 3-minute key. If you have dual core, the PBX will worst-case block only one core, which leaves the other core available to log in and check what is going on. If you cannot log in, or the server does not respond to PING any more, then there is something very serious going onand I would say low probability it has something to do with the application (PBX). If you can log in, take a look at the taskmanager. If the PBX consumes the CPU or the memory, obviously then the one responsible has been found and we need to dig depper why this happens. The PAC example was something typical on what can be the problem and then it is releatively easy to fix it. We have also seen other cases where a device driver freaked out and killed the machine. The fact that a software is a device driver does not imply that it is best quality...

In Windows, you need to change something in the registry (see http://wiki.pbxnsip.com/index.php/Installing_in_Windows). Look for http://support.microsoft.com/kb/248611/en-us on this page. On other OS, the PBX might not have sufficient permissions.

This is almost a "FAQ"... What firmware are you using? I was under the impression that newer firmware versions fix that problem.

What about stuff like hot desking? Does it go to voicemail?

We will focus on the whole MoH/paging topic soon anyway. Then we'll see what is the problem there.

Funny... Maybe it helps to increase our position in the google ranking ! Maybe we should just include the robots.txt in the default deployment. Do you have by chance something that we can put in there?

What is the setting in the provisioning section of the domain? Also, are the users aware that they need to perform 11-digit dialling (1xxxxxxxxxx) for outside calls? No need to prefix with something like "9" for an "outbound line" like in the old days. THe phone needs to "1" to tell internal numbers from external numbers.

Maybe we can take the insecure offer out. I believe this was for an old firmware version of a phone, today it should not be relevant any more. What OS are you on?

Yes. pbxnsip even takes advantages of Vista, e.g. the full IPv6 support.

I assume that there is only one email address in the settings, so I am not touching that topic. The problem is probably redirection. Can you check what redirection settings are associated with the extension? Maybe there is a redirection to an assistant which is triggering the double emails.