Vodia PBX

You mean the same email twice? Or one email for the missed call and another one for the email CDR?

Sorry that was just the forum editing tool that tricked me. The text should be just username@domain, nothing around it.

Are you using the button "dnd" for this with the mass provisioning?

I just disposed my private router collection. I tried to make a statistics on whoich router is using what type; but in the end the result was: Every router as it's own type. There is a interesting IETF document at http://tools.ietf.org/html/draft-jennings-...test-results-04 that tries to classify routers. IMHO just another reason to finally give up on NAT and move straight ahead to IPv6. Or use TLS and encrypt the traffic between the phone and the PBX so that no "smart" routers can mess the SIP packets up! Security against vendors that want to do you something good, but unfortunately are unable to do so from their skill set.

Whow... I can't try this out right now... That would explain the problems. Does that apply only to "older" bootloaders or also the the latest and greatest? Can someone verify?

DMZ - are you trying to provision the phone from the "outside world" or from the LAN? The PBX has to generate a lot of links for the various files that have to be downloaded during the provisioning process. If it just puts 172.16.x.x there and the phone is somewhere at home behind a router that address cannot be resolved and we are talking about a tricky routing setup here. That is just the PIN code if you want to switch from user mode to admin mode on the phone. Has nothing to do with the provisioning itself. The username must identify your account. if you have multiple domains, you must use the form user@domain; if you have just one domain then just user may also work (play safe and use the user@domain form). The password is the web password of the account; or you can put the domain provisioning password there. Play safe and use the user web password here... If you have an authorization problem, then you should focus on the right username and password. Also if you plan to use DMZ, then try it out first without DMZ in the LAN. Don't place this file anywhere. Also if you have a pnp.xml file, move it away. Also for the sake of getting it working, move PnP files away that you might have in the html directory. After you get it working, you can move them back one by one; then you might see which one causes the grief. Sorry for all this trouble; but the 7.3.14 firmware is a little bit "bitchy" if you want to provision a phone without major security holes.

If the caller wants to call an extension, it is the best choice to use a extension for this on the IP-PBX. Then all the features that do exist for this application work nicely. The only workaround is that you cannot use a "native" SIP phone, but use a PSTN gateway to call the extension on a legacy PBX. So far so good. The first catch is the setup. If you have hundreds of extensions, setting that up can be a major problem. But if you take a look at the XML files in the working directories it should be relatively easy to copy them into other extensions with a shell script. The other catch might be licensing. If you have 600 extensions then that solution can become more expensive that a single trunk setup. But I would say this is a business problem, and you should try to make a special deal for this case.

It could be that the dial plan that you assigned to the agent group does not permit to place such a call.

Nope, the only workaround is the copy & paste from the web interface and re-format it to CSV format. The other possibility would be to clone the whole domain and then after restoring on another server delete what you don't want to carry over.

Yes this also works. In this environment you must use the username@domain form for the account. The password can either be the web password for that user or the domain provisioning password (if set). The URL that you have to put into the phone must be in the form http://pbx-address/provisioning/snomxxx.htm (no MAC address).

It is unbelieveable what kind of devices are out there in the router market. I have seen routers that have no expiration of NAT ports, and they just use 32 ports in a FIFO fashion. That router in a busy office was a nightmare troubleshooting why some phone calls just wont work!!!

You are right. DID are a problem. Because they are global, they get lost when you want to restore them in a domain.

Hmm. You have a point here. Maybe the way the MoH works right now should be changed. The MoH source itself should already insert the messages, instead of having the ACD do that. Then you could pick the stream that you want for the occasion that you want. Would not be too hard (I believe). But would be a change that is not backward compatible and that is always causing an uproar.

Right. If you do HTTP authentication then there is no more need for the MAC. The only point of the MAC is to group extensions, so that it is possible to provision multiple extensions on the same handset. Only if you want to do that you need to put the MAC into the extensions that should be provisoned together.

Not yet. But is a core component of the next major release.

I would either clone the whole configuration of the PBX (if it was a single domain setup) or get a domain backup and then restore the config on the new server or create a new domain from the domain backup. Then after the restoration, delete what is not needed any more.

snom: 7.3.14 Polycom: 3.1.2

Yea, VLAN setup is still very tricky. The problem is that when the phone boots up (after a factory reset), it is not in any VLAN and needs to get there. If there would be a way to tell the phone to switch into a VLAN then things would be straight forward. Maybe LLDP is the answer?

Can you upgrade one phone to 7.3.14 and see if that changes the picture?

In 3.3 we introduced the "EPID" (endpoint identifier). When someone calls 911 on a emergency trunk, then the PBX will put the EPID into the contact. When the emergency center calls back, the call goes directly to the extension. even if there is usually an auto attendant in the middle. This goes beyond what good old TDM was able to do. VoIP and SIP can do more than emulating circuits.

Not yet, we are still collecting items for 4.0!

What country are you in and what carrier are you using?

Unfortunately, presence is not presence. Microsoft uses their own way of presenting presence, which is AFAIK now compatible to the presence that counterpath is using. The PBX in the middle does not translate the presence, it does not even generate the presence. As OCS is becoming more popular, we'll have to see if the PBX can "synthesize" presence information and pass it on to OCS.

From the CPU perspective there is no difference between occasional and permanent recording. You don't want to have stuttering voice when a user records a couple of seconds; and you can never know if several users press the recording button at the same time. I would look into a new system...

Yea, we need a code for "retrieve my call that is connected on a trunk" (instead of a call that is currently on hold or parked) and we need another code for "blind transfer my currently active call to the cell phone".