Vodia PBX

Whow! IMHO you dont have to blacklist too long as the total impact to the system performance is really getting very low if you blacklist for something like 7 days. Anyway. The dropdown just contains some useful (as we thought) proposals, you can edit the page in the admin/email/texts section and add another option for example for 365 days or longer.

Is that English? Well the answer to this one is that the admin should be able to edit every page. The PBX just comes just with the templates. I believe for the emails thats very obvious, but also for all other pages it makes customization very flexible.

In the extension we have the "Title", kind of a note pad... Though it shows up in the missed calls field!

The problem with the PSTN is that it really depends a lot on the location where you are (for example, ISDN country) and what kind of termination you want (for example, T1, E1, J1, 1 x FXO, 2 x FXO, 4 x FXO, 6 x FXO, 8 x FXO, 1 x BRI, 2 x BRI, and so on). Even FXO differes from country to country with the impedance and the caller-ID. Picking one is not enough.

Not really. IMHO it is not top priority because there are so many different gateways, you install it only once per site and users don't have access to screw up the gateway (in contrast to the phone on the desk).

Who-how! Automatic blacklisting seems a killer feature these days!!!

Is there any chance you can try another service provider or a PSTN gateway, just for the sake of nailing down where the problem is? I dont think that hangup depends on the codec type.

Well usually it is a feature if the OS can swap other tasks to other threads to more cores, the only concern from the PBX perspective is jitter on the RTP playout. I consider it a feature that it locks only this one thread down. But for sure it is a matter of documentation, and this forum thread should help a lot understanding it!

Disconnects after 30 secs usually happen when there is a SIP routing problem. The ACK message probably does not make it. THis is typically the case if the service provider expects that the customers are running their clients on a public (routable) IP address and/or don't use a session border controller. In this case you either need to get a routable IP address for the PBX or switch to another service provider.

We have seens the following addresses over the past few weeks: 109.169.41.129 173.224.209.188 184.82.2.134 200.33.181.18 202.67.217.133 213.174.148.146 221.195.72.20 61.242.169.1 62.96.7.99 64.120.170.101 64.22.82.2 67.222.10.134 78.141.172.140 82.195.143.18 (a lot) 86.107.102.123 94.23.197.75

Not sure, better check with Sangoma.

In Linux the behavior is different than in Windows. The intention behind settings the affinity is to avoid RTP jitter during the time whne the OS moves the thread between the cores. In Linux, this is only relevant for the RTP thread, in Windows the whole process must be bound to one core. If you want to set the affinity in Linux, you should do this from outside of the process using the standard Linux command line tools (see taskset).

No, because those ports are allocated in a "random" fashion anyway. For DNS, the OS will pick a free port and for the RTP port, the PBX will try up to ten times to get a free port in the range specified.

RTP ports are always bound to 0.0.0.0 and ::. Also the DNS client sockets connect to 0.0.0.0 and ::. Because those ports are client ports and discard messages that dont match the SSRC or the DNS open query sequence number IMHO that should be okay.

I agree on that. There must be a soft phone in the package. So far there is the good old snom 360 softphone which still runs even on Windows 7. Maybe snom (we) should just take another photo of a newer phone model and generate a new soft phone image...

You actually talk SIP to a local IP address (typically 127.0.0.1) and the RTP also flows locally. Looks a little bit odd at first sight, but works nicely. I saw that also web cams start to use their own IP address locally, seems to become more and more popular. Also it makes it easier to design it for any operating system, no need for device drivers!

There are two topics: Media and signalling (SIP). As for media, it is usually safe to use to use loose RTP routing. The IETF defined something very NAT unfriendly with the usage of the RTP ports and usually no vendor is so strict. You can do this on trunk leven, but also for the whole system (in the admin/ports section). As for signalling, yea the other big topic is on how to "say who you are" when using trunks. The conflict is the indication of the caller-ID and the other thing the indication who pays the bill. This is a big mess when it comes to SIP trunking. IMHO the RFC says that the From-header contains the caller-ID that should be displayed on the screen, and the P-Preferred-Identity (or Asserted-Identity) says who will pay the bill. Unfortunaly there are is a lot of free software out there that thinks the From is the one who pays and then other headers like Remote-Party-ID will say what the caller-ID is. It gets even worse when service providers don't use session border controllers, because then it mixes even more when the SIP proxy and the PSTN gateway use different interpretations! That's why we added this drop down to give you several choices. I know it is bad, but you have to play with this setting until it works. The next release will make it even more flexible, because this is really a problematic point with the SIP trunking today.

So far (please post additional if we forgot something here): AudioCodes gateways Cisco gateways Patton gateways Mediatrix gateways Quintum gateways Vegastream gateways Beronet (Berofix), especially for ISDN AVM FritzBox (ISDN, may require a special firmware to acts as a PSTN gateway) Teles VoipBox Sangoma (PC card) Dialogic (PC card)

That looks good.

The name of the setting is a little bit misleading. This applies only for multicast RTP. If you want to bind e.g. the SIP UDP port to a specific IP address 12.23.34.45 port 5060 and 6.5.4.3 port 8000, then you should use the setting "12.23.34.45:5060 6.5.4.3:8000" where you have the port right now ("5060").

I am sorry? Maybe we have a different understanding about intellectual property and licensing a software product.

Come on. You want multiple domains with 10 extensions? The point is that snom ONE is sold through the VAR channel, and thats the reason for the CPE focus.

AFAIK you can do that already today. Sangoma offers a SIP interface to their hardware which works with pbxnsip/snom ONE.

On a single server, definitevely not. There are a couple of ITSP running the PBX in hosted mode. The biggest challenge in doing so is the transition from managing one server/core with multiple domains to multiple servers/cores with multiple domains. Add virtualization to this, then you also have a quick failover solution. When those steps are made, the service can practically scaled endlessly; as long as domains stay smaller than lets say 100 or 200 users per domain you can always bring up new servers to deal with the increased load. What those ITSP that I know did was making the standard tasks like setting up domains, deleting domains, moving domains as automatic as possible using scripts. The main problems is the link to their CRM systems and database, and linking it to their billing solution. Because there are so many different out there, there is no standard solution available. Also, keep in mind the PBX is "class 5", not class 4. If you need a solution for dealing with a large number of subscribers without touching the media (except for the use of a SBC), then you should look for something like BroadSoft, Metaswitch or Telepo.

We have seen CardDAV. Nice standard, but who supports it? The topic remains messy. The "addressbook" as such does not exist, it is a scattered database with a lot of local copies in all kinds of formats. Therefore, there is no simple answer for this. The only common dominator that I spotted to far is vCard, so whatever we do in the future will probably try to use this format as much as possible.