Vodia PBX

Check the log. There must be something like "Trunk xxx sends call to yyy" (log level 5). The yyy must exist in the domain, otherwise the PBX will generate that error code.

I had the same problem with a Cisco 7961 phone. Maybe rejecting a phone call is not on the feature list in the Cisco family...

The create-button disappears when you select a specific MoH-source (then you can edit it). AFAIK is it not even possible to limit the number of MoH sources.

I would choose a DNS-routable domain name as the primary name for the domain. "localhost" is unusable in this case. Do you see that the PBX identifies a specific trunk? If that is the case, then check how the trunk determines the account. Maybe there is something wrong.

Hmm, so that you have more information about the call?

We already ran into the ## idea when calling back from the mailbox (that is a common feature when calling back from the mailbox). So far the idea to use ## when the cell phone got called is new. And we should also know what the use case it - sounds like you want to transfer a call to a someone else, and then the next question is on how that transfer should work.

If that is becomnig a common request we can add it.

At least we can say that it does not matter if you are using a CS410 or a PC-based solution... I guess you followed the explanations in http://wiki.pbxnsip.com/index.php/TAPI_Service_Provider?

That might be a problem. Maybe that usse agent tries to be smart and there is a router in the path that does not support (too) smart devices. In other words, don't use STUN. The TCP/TLS timeouts are probably okay, because the PBX initiates those connections and they are not kept alive with registrations.

Hmm. You should also upgrade to 3110 (the latest release), but honestly I am nore sure if that will fix the problem. I guess we have to re-test this in the lab.

Oh yea, maybe Cisco should consider supporting PnP ALG as well :-)

For HTTP the problem is that during the connect the PBX has no possibility to find out which domain the web client wants to connect to. That is not 100 % correct as there is a new RFC that allows the web client to tell the web server what domain certificate it would like to see. Anyway, the PBX does not support that yet. That is only for email and it is also a global setting.

I believe the easiest is to record some dummy prompts and then look into the file system, play them back and then replace them accordingly.

AFAIK OCS is strict on the certificates in TLS. The PBX uses just a default certificate, and probably OCS does not like it. In order to get things working I would start with TCP first. Then if that works you can try using TLS and you might have to import a valid certificate (and the respective private key).

Maybe you can make an example, I still don't get the point...

Did you switch to the TCP transport layer (see http://wiki.pbxnsip.com/index.php/Polycom, "Switching the transport layer")?

Though not very beautiful, you can do something. I would choose some additional digits of the code that are always the same :-P just to make sure that the caller does not enter complete nonsense. Then you can use a pattern like that: !123[0-9]{4}!300! BTW the pattern that you wanted to use probably looks like this: !1234|6789!300. The [] group alternative characters, that is probably not what you wanted. If you have not too many customers, that would be a pragmatic way to get the problem solved!

The PBX has a loop counter. If that exceeds a value (I think default is something like 10) then the call is treated as unroutable. That is good for the PBX health!

Nono, you can have as many as you like. Where did you get that limit from?

Well, the phones are supposed to send some kind of keep-alive traffic. How can the PBX tell if the phone did not reboot in the meantime? What phones, what version?

That's still my favourite: http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. The "Separation by Route" also applies to the generation of IP addresses for plug and play. Of course, it is not recommended and very support unfriendly. But it should work ...

Do you want to use an external application for modify the Caller-ID? Do you want to change the Display-Name or the telephone number? Maybe the address book could be an option for that.

I would use a agent group. If they want to see who else is calling there are several ways. Depending on the phones they can use buttons for that, or they can use the web interface to monitor calls in the queue (now nicely working with AJAX). Of course an agent can put a call on hold and make another call, for example, pick up someone from the queue with *87.

Maybe just SSH into the box and take a look around. Can you ping pool.ntp.org from there? Maybe something wrong with the DNS settings.

There are two things you can do: 1. You can record that in the personalized annoucement (*98 recording for the AA), and then switch the PBX dial by name annoucement off (in the IVR tab). That would stay local to the domain, even to the auto attendant. 2. You can replace the file in the file system with your own recording. That would then apply to all domains.