Vodia PBX

Now there is a beta2 image available at http://www.pbxnsip.com/download/pbxctrl-2.1.0.2090.exe (see http://wiki.pbxnsip.com/index.php/Installi...Manual_Upgrade) on howto upgrade to this image. Things that have been fixed were a memory leak in the subscriptions subsystem and a general review of the memory management, maybe fixing some (extremly rare) race conditions in the memory managament. There is also a new feature (*00) which calls the extensions cell phone directly without ringing the extension and without respecting the fork delay and day of time settings (so that you don't have to remember an extension's cell phone number any more). We will also provide an update for the CS410 in a seperate posting. We would like to get further feedback from the Windows platform users before we turn our attention to the Linux platforms.

Yes, talking about 7.2 of the snom firmware. I guess we need to update the Wiki on this - probably a bigger project. That's why I hope that this version comes out and everything will be fine.

The latest version (see post in this forum) uses AGC for this problem. And we keept the overall volume relatively low (that's ovciously what most analog handsets vendors do). That did the trick in our office, where we also had a nice long list of audio problems. Polarity change should be included in that version as well, so it is definitevely worth a try.

Limit: Number of accounts or extensions. Sometimes, when people start with demo keys, they can create accounts that are later not licensed any more when they get a permanent key. That's what I call over or at the limit.

In general, you can move an installation by copying the working directory of the PBX. It is also good for backup reasons. Then in the new location, you can make the neccessary changes. You should definitively take a look at http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. There you can also see what are the options to run the PBX so that you can access it from the outside and the inside of the office.

Some countries use polarity change to indicate disconnect. Maybe Sweden belongs to this group. Detecting tones is extremly buggy. If someone sings a song on the phone you might detect that as disconnect and hang up.

Well just saying that voicemail fails is not very specific. There are other places where it works, so there must be something different in your location. Ideas for differences: What version are you running? What Email-server are you using? Are you running the email over public Internet? Is there anything suspicious in the log file?

Well, with the snom we are in the middle of finishing something really nice. The 7.2 version will support "buttons", where the PBX can take full control over the LED. it will also support XML-based directory, where the phones pull the address book on the fly from the PBX. Unfortunately, there is no usable 7.2 version available yet...

Ping tells you that the OS is still alive. SNMP tells you that the application is still alive. Rebooting is not an attractive option. Maybe you are suffering from the memory leak in the head branch (2.1 beta), we are trying to isolate it. Check from time to time how much memory it allocated.

Well, but then you were already on the limit, maybe above it...

No... Supposed to be a feature! Why would someone want to hide the caller-ID for a call within the company?

Hmm... MP3 is a format that was designed for music. Maybe you can compress the audio into a different format? There are a lot of speech codecs available (e.g. GSM, G729, G726 and maybe iLBC). Or maybe it is just because the MP3 parameters don't fit well. For example, don't try to use stereo, and make sure that you have enough bits/s for the result. Maybe try with a 64 kbit/s compression and then go down.

Do the ulaw files sound okay when you play them back?

Well, AudioCodes definitevely works. We worked on a better international support in the latest version (posted in the cs410 forum), it is surprising how the different countries are using the analog wires. I would say it should work in Germany and Italy, possibly in UK.

The BPX writes the files in ulaw format because that is the least CPU intensive option. Once the file was written to the file system, the PBX does not need it any more and other applications can use it. There are a lot of audio tools available that can convert audio formats (e.g. http://www.goldwave.com). If you run them periodically, you can convert the audio formats "offline". I am not the big expert on how to run a problem periodically in Windows, but I have witnessed it is possible and maybe someone else on the forum can help out on how to do this.

Hmm. There was a similar bug when sending a call from the auto attendant to the calling card. Sounds like a bug to me. Will be fixed in 2.1.

Well, that is the question of the PBX gateway. Maybe the PSTN gateway accepts URI like sip:*82,12345678@domain, then it could insert a pause. But there is no pause sending out the INVITE (well it is one packet). But I don't see a reason why the service provider would need a pause. If you can put such a number on a speed dial of a analog phone, it should also work.

Any change to use SNMP to monitor the uptime of the box?

We are not talking about two-stage dialling here?

When using exactly that dial plan, I get: [5] 2007/08/20 14:01:20: SIP Tr udp:192.168.8.118:5060: INVITE sip:*2123653222@192.168.8.118 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK-80b94d59b29bc3aa6ca9c0e7fb5297f2;rport From: "Susi Sorglos" <sip:41@laptop.pbxnsip.com>;tag=50395 To: <sip:*2123653222@192.168.8.118> Call-ID: 73673963@pbx CSeq: 20439 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.2.100:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.0.2085 Content-Type: application/sdp Content-Length: 274

If they share mailboxes then they receive MWI indications from both the individual and the shared mailbox. I think that could explain why the phone "mixes" the status messages. Workaround could be to disable the personal mailbox or ask them to dial into the shared mailbox and enter the PIN for the shared mailbox (no MWI for the shared mailbox).

The Polycoms have a special flag for an address book entry that tell the phone to subscribe for the dialog state for that entry. This way, you can monitor the state of that extension or CO-line.

What version? Any chance to move to the 2.1 version?

I got mine working at some point (see http://wiki.pbxnsip.com/index.php/Nokia). It is really not intuitive. But possible. Make sure that you have the version 3 on the phone. I went to my Nokia shop, they gave me an upgrade from 1.x to 3.x and that was a big step forward.

Hmm. I just tried it here and here it was okay. Maybe a problem with space characters?