Vodia PBX

Done. Also snom is learning...

You mean http://forum.pbxnsip.com/index.php?showtopic=1312? 3.1 is the "next build"...

There is no limit. Though 300 sounds like a lot to me. Usually it is possible to use expressions to make the list shorter.

Errno 22 is "Invalid Argument" - seems like the timeout for select() got screwed up in that case. "Should not happen" - but it seems we need to put an extra safety belt on here.

First, it should also be possible to lock the processor to a core from outside of the program itself. Seems http://www.cyberciti.biz/tips/setting-proc...or-process.html is a interesting link to do that. Maybe the apt-get also installs the neccessary stuff so that the processor can do it on its own. I think the link above is interesting reading regarding this topic.

Not all countries allow recording of calls. So we definitevely have to have a license key for recording, even user-initiated.

For those who can live without extensive release notes, there are new images for 3.1 out: http://pbxnsip.com/download/pbxctrl-3.1.0.3043.exe (Windows, manual upgrade) http://pbxnsip.com/download/pbx3.1.0.3043.exe (full Windows installer) http://pbxnsip.com/download/pbxctrl-freebsd7.0-3.1.0.3043 (FreeBSD7.0 32-bit) http://pbxnsip.com/download/pbxctrl-rhes4-3.1.0.3043 (RedHat 4) http://pbxnsip.com/download/pbxctrl-centos5-3.1.0.3043 (CentOS 5) http://pbxnsip.com/download/pbxctrl-suse10-3.1.0.3043 (SuSE10) http://pbxnsip.com/download/pbxctrl-debian4.0-3.1.0.3043 (Debian 4) http://pbxnsip.com/cs410/update-3.1.0.3043.tgz (CS410) We'll have the release notes and the missing OS (e.g. MacOS) ready probably on Tuesday, until then please drive carefully!

:-S the workaround at this point is to record always the following message together with the morning/afternoon/happy Xmas...

Oh so you want to have a pre-welcome greeting?

Maybe I don't understand this... There is already a IVR tab for the auto attendant where you can specify depending on service flags what welcome greeting should be played back.

If you like give http://pbxnsip.com/protect/pbxctrl-3.1.0.3042.exe a shot.

Well, you can set the "lines" parameter of the extension to "1". That means there is only one call going to that extension at a time.

Well, that's what the settings "timeout_conference" is good for. If a user-agent "sends" one-way audio (which is no audio) that is okay. There are many ways around it, for example choosing a user-agent that sends keep-alive RTP (silence indicators). But we want to compatible to devices that are not so smart. I would not change the general maximum durtion of the call. It is a uneccessary burden to the system.

Hmm. I remember there was a problem with those old servers. Might be fixed already. Maybe you can wait a few days and then try the upcoming 3.1.

Well that message itself is not a problem. It is actually the only way to find out if the server understands EHLO. Does the rest of the email delivery work?

Ops, good catch! 3.1 is hopefully becoming a reality very soon.......

Well that looks like your service provider disconnects the call... It is really strange after 11 minutes. Maybe they have a 11-minute demo key from their softswitch vendor ... No it seems that the re-INVITE fails. Looking into it... Can you send us a private mail with your account information so we can give it a try from here?

Well, it is a global setting (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File). There is also a small trick to set the setting through the web interface. It is not visible through ther web pages because we usually don't want people play with these settings.............

Who disconnects the call - the service provider or the PBX? (Does the PBX send the BYE request or the "200 Ok" response?)

After the maximum call duration, the call gets disconnected anyway. The default is two hours. You can make this setting shorter if you like; though I would believe it is a resonable setting. There is also a "hidden" setting called "timeout_conference" which is by default 3600 seconds. This value was so high because there are user-agents out there that do not send keep-alive traffic when mute-ing a call. If you have devices that do send keep-alive then it is safe to lower this value to something like 600 (10 minutes).

Actually, plug and play should work. Punch in the IP address into the "TFTP server" setting into the phone, and set the Admin->Settings->Ports->TFTP policy to always send the password and set at least one extensions MAC address to "*" and give it a try. Otherwise, just make sure that you fill in the outbound proxy on the phone and also use a domain name in the line settings.

What version are you on? Could be that we already changed that in the meantime.

Well, the SIP traffic can be seen through the web interface of the PBX (well, filtered out keep-alive stuff). The tcpdump is a tiny version, but maybe it does have the port option.

You can still override the OS routing mechanisms, and that also applies to stuff like TFTP (see http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses). A typical case is DMZ. And, of course, DMZ is a bad thing to do in SIP environment. It is a lot easier to have a routable IP address, a public IPv4 or IPv6 address.

tcpdump is the right command. Not sure about the "-i any", maybe you have to specify what interface you want to listen on.