Vodia PBX

Ops, yes. The domain pages and the user pages still had the old hardcoded extension. Will be fixed in the next release.

Yea the setup of the forum requires to check the permissions. Probably not all forums are the same. We changed the AA forum.

Yes, the auto attendant just wants to make it easier to enter extensions in envirnonments where the extension length is fixed. You can still use the option "user must hit pound" or "when the extension matches" (which is the default anyway).

Yea I would recommend to do a test setup and get an Ethereal trace. There must be something wrong with the negotiation of out of band DTMF.

We could add a parameter to the config files. For the time format, it might even make sense to make this dependent on the time zone (US = 12 hour format). I am not sure about call waiting, that can also be configured with the number of lines for that extension.

Well that is a motivation!

In 2.1.2 it is still hard coded to the domain address plus /help/help. In 2.1.5 there is a new setting for the help web page. Maybe a reason to move to 2.1.5!

Is there anything suspicious with memory usage or the number of threads?

Whow. Does the PSTN gateway negotiate out of band DTMF (should be visible in the initial INVITE coming to the PBX)?

ERE are not so simple... That is why we recommend to use the other "simple" forms - if possible.

Well, the PBX processes the flags from from to end. The first that "fires" is being processed. I am nore sure if it should be the first or the last, maybe it even depends on what you want to achieve. No, the service flags don't know about each other and work independently. Well, if you automatically change a flag by the PBX and then let the user change it manually, there is always the problem of secretaries leaving late and comnig early and then the PBX overwrites their manual state in a way that they did not envision.

Well, all these codes have practically no meaning and the operators use them as they like and might even change their mind. Sometimes the codes even depend on the termination providers down the SIP trunking value chain... The only difference is the code class (4xx, 5xx or 6xx). So much for the defense of the current setup...

Do you see the alert-info headers in the SIP packet? If you configured the Polycom by yourself, you probably need to change the sip.cfg config file to pick up the right melody.

I would recommend to track the registration stability by email. In 2.1.3/4/5 you can let the PBX send an email out when the registration status of an extension changes. That was extremly useful to track registration problems down in other installations. There can be a lot of reasons. If you can, try a different router. Some routers tend to get instable if there are too many UDP ports open.

There is a RFC for this (which is supported by the PBX). AFAIK Polycom does not support this yet. http://wiki.pbxnsip.com/index.php/Indicati...ge_of_Caller-ID

At the moment, you can have two calls on PSTN ring only through static registrations where the PBX believes this is a call to a SIP phone. The problem with PSTN is that stuff like trunk failover also must work. Early media can also not be supported. Forking calls to cell phones will mean that if a cell phone looses its registration, the cell phone mailbox will connect the call. This makes the PSTN fork limited use to me anyway.

Well, what codec is eyebeam using? I think it is using a different codec than the PBX, and you probably end up in a transcoding situation. If you do transcoding between different low-rate codecs, don't be surprised if it sounds like in the bath tube.

Okay, maybe yoiu should try the pattern ^1([0-9]{2})@.* then. The means that it should match the beginning of the string.

Do you have a comparision number of CPU performance if you turn the real-time scanning on? I think it could be an important CPU performance difference. Also, I heared that once you installed PCAP (the Wireshark subsystem), it eats CPU no matter if you run Wireshark or not. This is because it puts itself somewhere in the driver stack and processes every packet. It is a rumor, maybe someone has more information about that.

Also, the 2.1.5 version has more logging on how it processes the dial plan in log level 9. Just in case you need to understand what the PBX is doing when things are going wrong with the dial plan.

It does not matter what page you are loading (reg_status or reg_settings). The important part is what you put behind the ?. It is like filling out a form. Because the web interface has no form for pass through mode we need to use this little bit inconvenient way of saving the setting.

Well if the phone uses a contact of 0.0.0.0 then there is probably something wrong with the phone.

Maybe worth a try: set allow_pass_through to false. (e.g. http://192.168.1.2/reg_settings.htm?save=s..._through=false).

In the auto attendant, you can exclude certain extensions ("Accounts that cannot be called"), this also applies to the dial by name feature. Bob and Robert is difficult. If you want to keep his last name, you have to decide which name you want to take...

Affinity: the default is that the process uses CPU 1. I guess you are not using a virtual machine, where CPU affinity becomes "relative"... It would be interesting if the call is in pass-through mode. Then all packets should be forwarded "immediately", this should actually reduc ethe jitter compared to previous versions.