Vodia PBX

I heared that practically only ACME and Nextone are left in the SBC arena. They are hardware based. As far as I can tell they are both pretty good. Don't expect many blow's and whistles, but these devices are rock solid and that is what you expect from a SBC. Interestingly, the do also RTP relay. On other words, once that the RTP makes it to the SBC, it is only a ms away from the PBX which makes the whole RTP relay discussion quite pointless. Jasomi tried to do "media path optimization", but IMHO in the real life the environments were so difficult that this did not work in all cases. Maybe we will find out that the good old class4/class5 architecture was not so bad. With the difference that this time, the Centrex is an economical interesting alternative to the CPE-based IP-PBX.

Strange. What raises my eyebrow is that the domain index is "2" - do you have more than one domain? If the PBX sends the call to the wrong domain it is no surprise that the PBX does not find the address book there. Also, make sure that you don't use the NANPA dialplan in the domain. Your numbers start with "0" and that does not look like a U.S. number.

Well, the trouble ticket procedure is described under http://wiki.pbxnsip.com/index.php/Trouble_Ticket_Processing. But there are a couple of people who can help you directly without opening a trouble ticket. Maybe we should list them somewhere.

If you can please try http://www.pbxnsip.com/download/pbxctrl-2.0.9.2059.exe (see http://wiki.pbxnsip.com/index.php/Installi...Manual_Upgrade).

Maybe you have a gain problem. If the gain is too high you would hear static cross-talking. Try reducing the gain and see if the effect goes away.

I would give that a shot.

$-) But we really need a license solution for park orbits and also for idle hot desks.

If there is an option, turn voice activity detection (VAD) off on the gateway.

Well the point is that the box was designed for users that would use 4 "lines" - if those lines are running on SIP trunks then that is fine. Just don't expect you can run hundreds of call center agents on an embedded platform like this. There are still PC available with dedicated I/O subsystem, multiple core GHz processor and so on. Once you have a server room, moving to PC-based solutions is definitevely an option. The beauty of the box is it's solid state (no fan) and size. Perfect for a small office. Dentists, lawyers, shops, plumbers, real estate agents and so on usually don't run server rooms...

Well, as a matter of fact most of it can be used for voicemail. At 2 KB per second, you can estimate that there is plenty of space for the voicemails of a small orgnaization. The number of calls depends on the type. A conference is different than a plain call. But as a rule of thumb the box is able to to handle 8-10 calls, which should again plenty - considering that only four of them can actually leave the organization through the FXO.

Yea, we have to update this. It would be great if you could help us with working files - we can then use them as templates.

We are talking here about a few weeks here. If you like you can download a beta version. We will post something in the annoucement forum later.

Happy to hear that.

It seems we fixed the issue in the latest version... 2.1 will have it.

What? They cannot deal with a URI like sip:123@sipgate.co.uk;line=12345? Or is it just that the domain is not known to them? What does the REGISTER message look like?

There are two ways to address this. The first is to generate links that initiate remote calls from the PBX. Those links are e.g. included in emails that are sent after missing a call. Unfortunately the Wiki is down at the moment (grrr), so I cannot show you the link. The second possiblity is to use the TSP (TAPI Service provider). This feature will be available in the 2.1 release. We are testing it and it seems to be "okay" (another link on the Wiki grrrr). The good thing about this is that the TAPI support of the address books of the world is generally great, and we can just use all these wonderful tools to fetch telephone numbers.

I am afraid "!E!100!" is also no the right answer for those people........ I would recommend using the auto attendant. The IVR node is really something advanced. If there is something missing in the auto attendant to make that possible lets think more in this direction.

Hmmm.... Do you have USA dial plan for the domain? If that is so, the PBX interprets the numbers in the "NANPA" style, meaning that numbers must be 10 digits and "international" numbers must start with 011. You can check the entry in the database, look at at the address book in the file system. There is a "display name" and a internal name that is used for comparisons. Maybe that name was converted to USA style. BTW interesting PSTN gateway. Any comments on the gateway?

Have you seen then working in a real environment? I have seen the specifications and have doubts if they would work in a heavy traffic environment. The other question is if the PBX should act as a device registered to a BroadSoft/Sylantro platform. IMHO a combination of BroadSoft/Sylantro as the heavy duty service provider platform and the PBX running on the "edge" (either customer premisis or at least close to the customer) is a interesting combination that solves a lot of problems. This would be quite similar like in the good old days. In that case, the SLA/BLA appearance does not really matter, as the PBX users will have their own appearance anyway.

:-/ Does *97 still work?

Workaround: Just record a short clip of silence?

Well you can have all phones on all groups. However, a "login" means that that agent is logged into all groups. The only workaround is that you use different extensions on the same phone and then you can use the extension's identity to log into a specific group. Next version will make it possible to see on the web screen who is logged in.

is "yyy.yyy.yyy.yyy" a domain on the PBX? The PBX compares the name in the first line of the SIP request with the domain names. If you use the domain name "localhost" it will match anything. We introduced that to make everybody's life easier.

Maybe when 2.1 is out it would be good timing to ask him again

Why are you using port 5061? I see that port 5060 should be used. ibm# host -t NAPTR sipgate.co.uk sipgate.co.uk has no NAPTR record ibm# host -t SRV _sips._tcp.sipgate.co.uk Host _sips._tcp.sipgate.co.uk not found: 3(NXDOMAIN) ibm# host -t SRV _sip._tcp.sipgate.co.uk Host _sip._tcp.sipgate.co.uk not found: 3(NXDOMAIN) ibm# host -t SRV _sip._udp.sipgate.co.uk _sip._udp.sipgate.co.uk has SRV record 0 0 5060 sipgate.co.uk. ibm# host sipgate.co.uk sipgate.co.uk has address 217.10.79.23 sipgate.co.uk mail is handled by 10 mail-in.netzquadrat.de.