Vodia PBX

You can use also programs that are able to stream RTP (e.g. VLC www.videolan.org). In the last few weeks we have optmized the PBX for this application, so that the player can send linear codec with any packet length. Mr X has the details.

The best way to do this is to change the template on the PBX.

Remember that snom ONE is not a gateway, it is (supposed to be) a PBX. Maybe there is a market for a SIP (Lync) to SIP (Service Provider Reality) gateway; however I would assume this is part of the Lync package?!

Not an easy task... I remember Matt Landis wanted to write something about this on his blog: http://windowspbx.blogspot.com/

If you have selected a country code, the dialplan will not see "+" symbols. Then the PBX will represent the number in the country code specific format (e.g. 011 in USA, 00 in ROW). If you dont have a country code set, you are right, the test area might be buggy and this needs to be fixed.

Another reason to release the password.

As far as I know, the port is 8722 (not sure why not 22).

The decision has been made that the password is a secret. I must say from a support perspective, this is an unsatisfactory answer.

We have had cases where the logging of the gateway filled up the gateway over time. If you can log in, check the disk space with "df".

That noise does sound like SRTP noise. There is another source, which comes from problems with the byte order; this problem could have been introduced in one of the last (unreleased) versions. I am not on top of what versions are out the, but I would say if you took a head snapshop three weeks ago that is probably the problem.

You mean Australia also stopped the daylight savings madness? <zone name="AUS1"> <description>Western Australia</description> <gmt_offset>28800</gmt_offset> <dst_offset>3600</dst_offset> <dst_start_day_of_week>1</dst_start_day_of_week> <dst_start_month>10</dst_start_month> <dst_start_time>02:00</dst_start_time> <dst_start_week_of_month>Last</dst_start_week_of_month> <dst_stop_day_of_week>1</dst_stop_day_of_week> <dst_stop_month>3</dst_stop_month> <dst_stop_time>02:00</dst_stop_time> <dst_stop_week_of_month>Last</dst_stop_week_of_month> </zone> Would become: <zone name="AUS1"> <description>Western Australia</description> <gmt_offset>28800</gmt_offset> </zone> What about the other timezones in Australia?

You mean when there is no phone in the conference, only calls from the gateway you have the same effect?! Is it white noise (loud thundering noise) or something with multiple of 100 Hz or 50 Hz?

You need to get into the web interface of the phone, then go to the identity (probably 1). Then go to RTP and turn RTP encryption off.

You can always edit the template for the web page from the webinterface (admin/templates). Then you can add another line with the value that you want in the HTML template.

Well it seems that the SRTP key exchange did not work as it should. You can try to turn SRTP off manually and see if that makes a difference.

You can redirect to an auto attendant... That is the classical case for the 0 option. Or what would a personal auto attendant do?!

The redirection is usually to an internal number. Workaround could be to redirect to a hunt group and then from there include an external number.

As long as your customers are not too smart to handcraft requests, you could edit the page in the web page templates page and take the setting out. Could be a short-term workaround. The PBX should pay a three-beep tone when the last coin goes down... However, for a whole domain that is almost pointless. A fix for this problem could be that you scan the domains directory with a script for the balance and send emails out when the domain credit is going below a certain level.

Unfortunately, the caller-ID representation is a mess in the SIP trunking world. There are at least as many ways to deal with the topic as there are service providers, it seems. Most service providers assume that you are using a softphone, and the caller-ID representation has no high priority there. One of the core problems is that for calls that are redirected from the PBX to the provider, that provider cannot be sure if the PBX is malware and wants to present a fake caller-ID. We proposed the use of the "Related-Caller-ID" header a few years ago, but it seems that this is not well-supported by the providers either.

Well, there is information on http://wiki.snomone.com. I would just start with a Windows PC, double click and read whats in the screen. If you want to dig deeper, you can install Wireshark; it has plenty of tools for RTP analysis. snom ONE also has information built-in, but more on the higher levels (MOS graph). Anyway, there should be plenty of information and things to try out for the upcoming long cold nights.

You mean on the phone? Not using the built-in provisioning mechanism. You can edit the templates from the web interface and hard-code a 2nd identity though, even the failover mechanism.

The original design was respecting the URI; the orignal rule was: if there was a user=phone in the URI, then the PBX would change the URI and replace the domain. However, dealing with the reality in the SIP trunking market, we gave a little bit up on that for now and assume that all calls practically come from the "PSTN" or at least from a virtual PSTN. So far we have not seens any practical uses where the URI domain name was really relevant.

Yea, the decision was that snom ONE does not support transfer on third party devices. Seems like snom police wanted that everybody is a good citizen and uses only snom products! But obviously that went too far. There is a product called snom ONE green which does allow this (although I can't find it on the product web site).

I believe findstr is the command you need to use. Maybe someone can help out with the exact way how to use it.

The problem is that only extensions are allowed to change the state of the service flag. It is not a problem with the calling card, this is a problem with the service flag. You can verify this by calling the calling card account from an extension and then change the service flag. I believe the only way to overcome is to allow a magic symbol "*" or even better "**" to allow that everyone is allowed to change the state.