Vodia PBX

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.

The source if the TFTP address can tell the PBX what interface to use. BTW there is nothing like an "internal" or "external" address. Think about a system that has ten IP addresses (LAN, WAN, VPN1, VPN2, WiFi, IPv6, ...). The OS routing table is responsible for determining what IP address to use.

It should do that (of course). It should be possible to provision internal and external phones through the same TFTP socket.

No, TLS is used to send out the INVITE through a secure channel. In SIP, you have to allocate the RTP ports when sending out the INVITE (unless you want to end up in an endless discussion about offer/answer). So it is a pretty expensive operation. Apart from that, I think if all 50 phones are ringing when someone calls in, not sure if that is what you want in an office.......

Is it really related to the MWI? Maybe it is something stupid like instable power? Kind of hard to believe SPA reboots on MWI events...

Hmm. Seems that the logic is screwed up again. That clarification for the explicit times was missing one negation - in other words the times you specify there are the times when the PBX would not call the cell phone. We'll turn it around, that will make life a lot easier.

Even if there is no RTP 50 INVITE are a challenge. Think about TLS, DNS, ENUM and all the blows & whistles that need to be taken care of. Plus people are waiting in line for address book lookups for all these 50 agents in their personal address book. Ouch!

Well, as said - the FAX recognition did work in our test environment. To me it looks more like a problem with the environment. Maybe the PSTN gateway does signal out of band, and then the PBX does not analyze FAX tones any more. Maybe try turning RFC2833 DTMF off on the gateway and tell the PBX to perform inband detection (admin/settings). And also set the codec preference to prefer ulaw, maybe there is a problem with alaw. If that works we can find out what we can do to use the out of band-detection cabability of the gateway to save CPU power.

I would change the preferences a little bit to 100 and 101 to make crystal clear what the preference is. But apart from that, that looks fine.

You mean "reset" like in "reboot, crash"? Does this happen out of the blue? Or is there a specific event, like an incoming phone call? Maybe it is just re-retching the configuration and for some reason thinks that it changed (which causes a reboot on the SPA devices).

Well, you can deal with that. But then you have to work with timeouts (in the AA, direct destinations). It works, but is not very beautiful.

Hmm. Does the PBX show the right time (e.g. in the log)? Maybe the domain is in a different time zone...

That is usually caused by routers that are not "ready for SIP" - their NAT implementation changes the ports. Or your refresh interval is too short and the router already closes the NAT binding. The result is that intermittently you cannot call the extension. It is a serious problem and you should fix this either by changing the router or by making the interval short enough for these routers.

Oh, so you are using multiple domains on the system? I assume that the inbound traffic does not have the domain name in the request?

This is a case of having seperate domains that want to call each other. Although they are located in different states, the case is not so much different than having the domains in the same data center, but on different CPU. First of all, you probably need two trunks in gateway mode. Both of them should trust each other, e.g. turn the accept redirect on (that should help with the question 2 below). I would stay away from prefix 1, it has a lot of problems (1xxxxxxxxxx being one of them). Better choose something in the 3xx-6xx area, if the business cards are not printed yet... You can use pattern 3xx and no replacement to route calls that have three digits. If you want to make a routing entry for x11 then you can just give that one a lower priority. You need to set the setting "Assume that call comes from user" for that. Then the PBX can use the dial plan of that extension, and it can also charge that extension. Hmm. Ideas are: A. Use the direct destinations (if there is still space available). B. Set up "ghost" extensions with static registrations that point to the other system, have no mailbox and an impossible-to-guess password. Not very beautiful, but that might solve the problem if there are noo many extensions in the other location.

Hmm. That seems to be a problem only for the SPA942; other phones don't have that problem, right? Any chance to get more insight? Maybe you can the "Log Watch List (IP)" in the Logging to the IP of the phone and monitor the traffic. Does the counter change or the "Messages-Waiting" (yes/no)?