Vodia PBX
-
Posts
11,110 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
I would prefer Monday dinner.
Does anybody have an idea where we could go? I only remember a sports bar in the staples center, could be a fallback-solution. Maybe someone knows LA a little bit better than me...
-
Least idle?? Most idle would be my guess. Or stuff like LIFO.
-
:-/
Are you talking about the Diversion header usage? AFAIK that should have not changed - unless you use the "charge" field in the trunk that goes to the Exchange.
-
In theory it should be possible with the latest 2.1 version after a call has been initiated to switch to video. This is just like T.38 - first audio session and then switch to "something else" - which can be video, T.38 or whatever. Worth a try, but I never tried it.
-
Here we go with the latest:
http://www.pbxnsip.com/download/pbxctrl-2.1.0.2097.exe (raw Windows executable)
http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2097 (CS410 appliance)
http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2097 (Debian 3.1)
http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2097 (SuSE10/32-bit)
http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2097 (RedHat ES4/32-bit)
There were still problems with one-way audio and unwanted Re-INVITEs.
-
That will be covered in the 2.1 documentation - once that it is finished...
-
It is a good idea to keep numbers in a normalized form. If you are in the USA reagion (10 digits fixed length), you should tell the PBX so and select the dial plan in the domain accordingly. Outside of USA people start to believe that the number +xxxxx (+ sign indicating that it is a globally routable number), I am amongst them.
Also that means that you should program the PSTN gateway so that it sends diallable numbers. Same in address book.
BTW because of this it is a bad idea to put prefix into the dial plan. At least for "normal" calls they should be diallable without a prefix.
-
Try the patterns
75xx
or
75([0-9]{2})@.*
In the replacement, try:
sip:*75\1@\r;user=phone
or
sip:\*75\1@\r;user=phone
(not sure if the escape symbol is needed)
-
The PBX probably sends out a MWI whenever there was a message recorded for that account. BTW it also sends out messages from time time time (around one hour) so that lost messages get recovered.
It should not have any negative side effect as the message count does not change and the UA should realize this.
-
Yepp, this should already be fixed, if you can try the latest RC1 image.
-
Yea, there was a lot of trouble with SRTP bugs.
In the 2.0.3 version we decided that we don't support any "backward-compatible" bugs any more and to plain (and possibly painful) SRTP.
The workaround is obviously to disable SRTP on the snom phones.
-
Yes, 2.1 RC1 has it.
-
Whow! That finding is very cool!!! That could explain why we sometime see (even on very powerful machines) jitter coming up as a problem.
We will add a setting for 2.1 so that this can be set up as a setting.
Seems that Linux also supports this, and it is probably not a bad idea to include it there as well.
-
Hmm. Does the phone send a redirect message to the PBX?
You can try changing the mailbox pickup timeout to 15 seconds, so that the PBX will redirect the call to the mailbox, not the phone.
-
You need a phone that uses SIP as interconnect protocol. Or you can use a ATA, but then you will have limited functionality.
-
Aastra also has "BLF" settings that we got working (at least the lights were changing state). I would try manual configuration here. You can also program speed dial buttons, which come handy for call pickup (e.g. program it to *87123 if you are watching extension 123).
Cisco: IMHO they don't support that yet in SIP. Skinny seems to be more important to them than interoprability.
Grandstream: Not sure. But AFAIK their extension board also supports dialog state, which is what we are doing. Manual configuration.
Polycom is a little bit more complicated as it is practically impossible to manually configure it (well, maybe worth a try), see http://wiki.pbxnsip.com/index.php/Polycom. There you need to use the address book and mark the entries that you want to watch. If you enter that in the extensions "Watch the calls of the following extensions" the PBX will automatically provision that for you. We did not figure out how to make them pickup a ringing call, but at least you can see that another extension is busy.
Snom: If you can live without Plug and Play, take a look at http://wiki.pbxnsip.com/index.php/Snom, there you can see how to use it with snom phones.
-
<email_from>"Phone System"<dallred@vensureinc.com></email_from>
If you are using 2.0.3.1715, better use a from header in the form dallred@vensureinc.com. There was some trouble with the parsing...
If you telnet to that server on port 25, you should see a welcome message coming. If that is not the case there is a problem with the SMTP server.
-
So there is no echo cancellation?
There is echo cancellation, done by a DSP subsystem.
-
2/10 means that two agents are added to the list of ringing agents every ten seconds.
The selection is random - amongst the agents that are not having a phone conversation on the PBX or are in the "recovery time".
-
Nono. If you open the box you will see that the chipset actually is mindspeed (www.mindspeed.com). They provide the audio subsystem (they have their own DSP unit for this on the chip!), and AFAIK they sold truckloads of these chips already. The problem with FXO is finding the right gain, and once that is done echo compensation is not such a big problem. We upgraded our software to the 2084 version and so far echo was reasonable.
-
No BLA yet - though the Aastra phones seem to be able to do that.
Maybe you can tweak the files a little bit ;-)
-
That should already work, there is even a RFC for that (Reason-header, RFC 3326). AFAIK snom phones should support that for some time.
-
Well well well.
SIP does not support SLA or SCA. That is sad, but true. The honesties in the IETF believe that "nobody" would need such a stupid, old-fashioned functionality.
So we have to live with workarounds. The workaround is "dialog state", so that someone can watch the status of an existing call. The PBX uses that for monitoring trunk lines (which we call CO-lines). You can see that someone is on the call, and you can see that it is "ringing" and if your SIP phone supports also this, you may pick up the call (using the Replaces header in SIP). When the call is parked, the PBX sends an updated XML as well that indicates a special flag that the dialog is not "rendering media" (haha). Essentially they developed a protocol for replicating the calls database. You can do the same watching with extensions and also with ACD and hunt groups. Interestingly you can do that also with night mode flags (although there is no call). I think we have done what is technically possible.
But it is all too complicated. Actually, the whole XML serialization stuff kills the CPU of the PBX and also of the phone.
We are working on a new package called "buttons" (outside of the IETF to spare us the endless discussions there), which is just used to turn buttons on and off. There is even interest from other parties like Asterisk guys. The idea is very simple: If the PBX wants to turn a "button" on, it just tells the phone to do so. No big XML hassle, just a small ASCII IM message and we are all set. This already works fine with everything that we could imagine, with the last piece of puzzle being the seizing of the shared line and the association of calls later. This protocol is also very suitable for software switch-boards that suddenly become quite lightweight.
So my recommendation for today is to stick to the dialog state. With that you can at least see what calls are going on in the office and that is the main point.
-
If you put them into the html directory of the PBX (relative to the working directory), the PBX will use these files as templates for generating the config information on the fly. There is some information in http://wiki.pbxnsip.com/index.php/Automatic_Provisioning, not 100 % up to the latest and greatest but a good start (we need to update that eventually).
Usually you don't have to do anything with these files - as long as you are satisfied with the default config in there. But we found that it is much more flexible that we hand out these files, minor adjustments become much easier this way. In that case - just put them into them html directory and change them as you like.
The latest 2.1 RC1 version contains all PnP files in the html directory, including those Aastra files.
RC2 & Exchange
in Microsoft Exchange
Posted
Ops, yea probably a upgrade issue as the PBX now differentiates between P-Asserted-Identity and P-Preferred-Identity.