Vodia PBX

You have essentially two choices here. First choice is to put everything into one box (e.g. snom ONE mini), then you have to use Sangoma cards with NetBorder Express. The second choice is to use a seperate gateway, then you have two boxes, but full choice in term of gateways. Especially in the (German) ISDN that can be an advantage, because there are a lot of local extensions for ISDN. Even devices like the AVM Fritzbox can be used (http://kiwi.pbxnsip.com/index.php/AVM), however the poor echo compensation at that time suggest to take a product with explicit echo cancellation (e.g. BeroNet, Patton).

I guess you have already seen http://wiki.snomone.com/index.php?title=Plug_and_Play. VPN will usually not work with multicase PnP, as most VPN routers dont pass the multicast messages through AFAIK. If you can, use something option 66 on DHCPv4.

Ja das sieht in der Tat nach einen NAT Problem aus. Im Wiki steht was zu dem Thema (http://wiki.snomone.com/index.php?title=Server_Behind_NAT), leider nur auf Englisch. Für einen stabilen Betrieb würde ich empfehlen, eine feste public IP zu verwenden, denn die ganzen Tricks mit STUN & Co sind in der Praxis doch eher instabil.

Wir haben mal http://wiki.snomone.com/index.php?title=Sipgate auf den 4.5-Stand gebracht. Damit sollte es eigentlich sauber funken.

The only thing that sticks out to me is that you have only one CO line. But that would not explain 66 seconds. I assume the PBX is behind a NAT router? This is the #1 reason for trouble... Do you have more than one IPv4 address on the host? Maybe you can just filter out the SIP traffic and send a link in a private message to me and then we can take a look.

No, it is always or never. IMHO it is better that way, otherwise it could be easily overlooked e.g. on holidays.

I dont think it will be a IIS issue with the limitations on how many requests you can have. The PBX sends the requests one by one. I would at this point really look at whats going on on the IP level, AKA Wireshark.

Das klingt für mich immer noch nach einem Problem auf der Signalisierungsseite. Wenn die Agent Gruppe den Anruf annimmt, wird der Anruf (normalerweise) verbunden. Es könnte z.B. sein dass das Gateway (welches? ISDN?) nach dem Auflegen den Besetzt-Ton abspielt der Anruf aber noch verbunden ist. Klar wäre es ein Workaround die Agenten-Gruppe zu umgehen, aber dann gibt es sicher später wieder Probleme mit anderen Anruftypen, z.B. Mailbox. Daher würde ich schon versuchen, diese Angelegenheit sauber zu lösen. Ein Log mit den SIP-Nachrichten würde hier helfen. Dann könnten wir sehen, ob das Gateway tatsächlich ein BYE schickt und die PBX trotzdem das Telefon weiter klingeln läßt.

It is a little difficult to tell from the code what is wrong. I would use Wireshark and look at the message on TCP level that are being exchanged. A working example can be found on the old Wiki at http://kiwi.pbxnsip.com/index.php/Linking_External_Application_Server_to_an_IVR_Node. I am not sure if that kind of documentation make it to the new wiki (I did not find anything).

There are a couple of links available from the pbxnsip web site about it. http://kiwi.pbxnsip.com/index.php/UHLL_Interface http://www.pbxnsip.com/phone-system/pbxnsip-hospitality-ip-pbx.php http://www.blogpbxnsip.com/2009/07/pbxnsip-uhll-license.html The technology has not changed since then. However, the documentation was not ported to the new snom ONE wiki yet.

No that should be all done through the web interface of the PBX. You should use plug and play to get this working; if you manually configure things on the phone that will be difficult; there are a couple of settings that need to be set to make it happen.

Dont be fooled by the name "Beta Corondidis". This is not a beta version, this is the name of a comet. We are happy with 4.5 so far; the biggest hickup was the upgrade of the trunks, where some manual work might be required with the trunk settings.

We have a UHLL middleware that can be used to talk to the hotel software. AFAIK many PMS are compatible with UHLL.

Das klingt nach einem grundlegendem Problem. Ich vermute dass das Problem auch bei anderen Anruftypen besteht? Sind die Telefone im LAN oder über das Internet verbunden? Welches PSTN-Gateway wird verwendet?

Hmm. Es gibt einige Posts hier im Forum zum Thema "AudioCodes". Ich bin mir nicht sicher ob das MP112 genauso wie die anderen Produkte funktionieren. Es ist aber eine tolle Kombination, vielleicht sollten wir das mal im Wiki aufnehmen wie man das Gateway mit snom ONE einfach aufsetzen kann.

Thats a pretty open question... Yes. The latest version supports REFER for all devices. No, that should work with anything. That was part of the problem . Nope. Just the general limitation on how many extensions you can have.

Das ist ein "normales" Problem beim Herstellen einer Vertrauensbeziehung. Im Log sollte stehen, welches Root CA Zertifikat in die PBX importiert werden muss damit die Beziehung stimmt. Wenn ich mich recht erinnere, könnte das Zertifikat auch bereits im Webinterface als abgewiesenes Zertifikat verfügbar sein.

Im web Interface unter Nebenstelle/Mailbox gibt es unten die Möglichkeit, Files hochzuladen. Das Format muß 8 Khz, mono WAV mit 128 kBit/s sein.

Soweit ich das überblicken kann, sollte das bei "sipgate Trunking" gehen. Bei eingehenden Anrufen must du "Ziel Konto" in der Leitung auf die Nebenstelle setzen wo der Anruf rauskommen soll; das ist jedenfalls der einfachste Fall. Ansonsten kannst Du auch einfach den Nebenstellen weitere Nummern zuweisen (z.B. "40 0211123456"), so dass die PBX bei eingehenden Anrufen den Anruf direkt der entsprechenden Nebenstelle zuweisen kann. Die ANI kann man ebenfalls auf Nebenstellen-Ebene definieren, so dass jede Nebenstelle eine andere Rufnummer darstellen kann.

I would make sure that internal calls work fine. Make a couple of calls from the 760 to another phone and check if they are okay. If thats the case the problem is related to the SIP trunk. There we would have to focus on NAT related problems with the firewall, as they are most of the time the problem.

That is obviously a problem of the phone. The 7xx series had a couple of software images over the past few months. Not sure which is the latest one, but you need a new image. That is a sign that there is trouble with one-way audio on hte trunk side. When the user disconnects the call, the PBX checks if the phone had to chance to hear anything, and in this case that seems not to be the case. It seems that your service provider does support the PBX running on a unroutable IP address, otherwise you would never get the INVITE from the SP. At first glance, it seems there is trouble with the firewall. Is this a cable modem router?

This is probably a problem with certain firmware versions of the phones. The TLS implementation is not all the same for the various models because of the different memory size, maybe you see a pattern with the models. I guess you have them all on 8.4 firmware versions? That could be another problem, although I know that there are installations with more than 30 connections on Windows:

What about just using a BLF type? I thought that mode would solve this problem.

Thats what the "Bind to specific IP address (IPv4)" setting is good for (in Admin/settings/Ports). You need to tell which IP addresses the PBX should send from, seperated by a space.

Huawei U8650