Vodia PBX

I guess that is something that we need to change on the PBX side... Or maybe we ask one of the participants to sing a song while waiting

You probably have to set up a gateway trunk. If the Asterisk server requires authentication, you can put the username/password into the trunk. You should definitevely specify an outbound proxy, so that the (pbxnsip) PBX can find out where the call comes from.

Ehh, that should not be neccessary. What firmware are you using? The other thing that really might make sense is to program the PoE switch to turn the phone off at certain times. Saving energy and as a side effect, make sure that the phones have no problem in the morning!

Take a look at this: http://kiwi.pbxnsip.com/index.php/Office_w...ic_IP_addresses. That should help getting this working properly.

In the Webinterface of the snom telephone, goto Advanced -> Behaviour -> Device Feature Key Synchronisation. Reboot required.

The latest and greatest can do it; however the web interface is not there yet and it would require fiddling in the XML files...

Seems like the tricky part is to become a user on that system?

I would check the XML files in the file system (grep is your friend here) to see if the content is what you would expect.

As soon as the phone indicates that it supports as-feature-event the PBX happily sense those CSTA messages.

Yes, has been done. Ask for a snapshot if you want to try it out.

When you check this flag the PBX will set a permanent cookie. Next time you want to login without a session, this cookie is used to bypass the login and go straight to the web page that is used after the login.

Well, I guess the people at the front desk would probably wonder how they can use soap to wake someone up... A better solution would be to use the web interface and set the wakeup from there. But you are right, we need to extend the wakeup call star code, so that you can put the extension number behind it (for example, *6240). The permission for that will be the permission to clean up the room, reset the extension which is typically used in hotel environments.

You can definitevely do this in an auto attendant (as the welcome greeting) or IVR node before you send the call into a ACD or hunt group. I believe newer versions also support that for ACD and conference rooms.

I cannot think of a simple way...

1283934474 = now - beginning of UNIX time... Next version will have it fixed. Until then, just ignore anything that is bigger than the max call duration.

Running the process on several cores is not possible. We tried, but this resulted in jitter in the 50 ms area which is not okay. Seems like the OS needs to put the whole process on hold (MMU) when creating a new thread that affects another core. What you could do is run two instances of the PBX in seperate folders and cores. This is like running two seperate PBX. The disadvantage is that you also need to maintain two PBX. If you are operating a server farm with many PBX and domains anyway, that should be easy. However if you just have one domain, this will be quite a big step. After all, it is a PBX, not a class 4 switch.

I guess you should take a look at the inbound routing of calls from trunks (e.g. http://kiwi.pbxnsip.com/index.php/Inbound_Calls_on_Trunk). Maybe in your trunk there is just the hunt group as the destination--and it would be no surprise that whatever you dial the call always lands on that hunt group.

Nope, that does not perform any DNS queries... You must use the dot-IP addresses (or colon if you are on IPv6).

This is defined in RFC 5373. Seems like the UA does not support this. What you can do is to add a line to the ringtones.xml file and define how the device would like to have it. ringtones.xml

Maybe just get us a PCAP trace (send an email to support). If you can (no big secrets with passwords), please include the working directory of your setup.

In Linux it should be relatively straightforward. You need two installation directories and do the proper setup. In Windows the service manager is the problem. By nature, it manages only one service. Though I heared about someone who was able to manage several instances.

OCS does not support UDP transport layer of SIP. Anyway, are you sure that you use the transport=tcp parameter in the outbound proxy of the trunk?

Question is what you want to achieve here. Generally it should be possible without such tricks. Maybe you need to look into the calling card account.

The old wiki is available here: http://kiwi.pbxnsip.com/index.php/TAPI_Service_Provider. This would also work in hosted environments.

As far as I remember version 3 the AA is a must (also in 4). When you call directly an extension you just want that extension to ring. Otherwise it would be difficult to program an address book entry from the cell phone if you always have to go two stages.