Vodia PBX

Please try http://www.pbxnsip.com/protect/protect/pbx...hes4-3.3.0.3147.

When the domain has the area code set, the PBX convers the incoming numbers into the +xxx format. You see that when you look at the call log from the system admin level. Then when calls are leaving the PBX, it converts the numbers back into human-readable format, or "carrier-readable" format. That's it. The dial plan also processes the numbers in human-readable format. No more worrying about numbers 1xxxxxxxxxx format or xxxxxxxxxx, or even +1xxxxxxxxxx (it is always xxxxxxxxxx, just like you have it on your cell phone if you are in the USA/Canada). There is a big difference between the "1" area code and all other area codes. Because the "1" area code means that numbers usually have 10 digits (exceptions being x11, 011 and 555 numbers). The PBX tries to be tolerant against different ways of representing regular 10-digit numbers, e.g. starting eith "1" and having 11 digits or 7-digits for local calls. For non-"1" area codes, the PBX assumes that international numbers start with 00, national numbers with 0 and local numbers 1-9. Extensions have less than 6 digits. If you don't want all the magic, you can leave the country code field empty. Then the PBX treats numbers just as strings without trying to be smart.

It does that already for the "IP Routing List". Maybe you can use it instead of the replacement list.

That sounds like the PBX believes it disconnected the call, the the phones don't. Then when they want to put the call off hold, they send a INVITE (a "Re-INVITE"), but for the PBX that looks like a new call. Also, callers in the queue are technically (from a SIP point of view) not on hold, they are "waiting" (connected). If all agents are busy, logged out, on DND or whatever, then the caller will stay in the queue, even if it is 40 minutes. The one luring out of the queue and not getting connected for two minutes because there are phones ringing and then hang up on the ringing call, well that call has a problem. It is like an agent presses the cancel button on that incoming call. What is the PBX supposed to do when a agent rejects a call? That is the question here. From the PBX perspective, there is no difference if a human being pushes a key on the phone or the phone itself rejects the call after a timeout. Anyway, the solution is to use the redirection feature to escalate the call after some time, maybe one minute. For example, the call can get escalated into another queue (priority queue), to a cell phone, to the boss extension, or even just into a mailbox. But keeping the phone ringing for 40 minutes does not sound like an option to me.

You need a 3.3 build for that. If you want to try - let us know what OS you prefer.

Whow, I wish they had the same attitude with the 79xx line...

http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses

I assume the problem is that SIP phones cannot ring forever. Usually the phone stops ringing after a minute. The PBX also has a maximum duration where it lets a UA ring. You'll see that when you look at the SIP trace. Either the phone sends a code like "487 Terminated" or the PBX sends a CANCEL to the phone. Having a phone ring "forever" is not a good solution. Imagine you are the caller and your destination lets you hear ringback tone for two minutes. I believe 99.9 % of the callers would already hang up after 30 seconds.

What does the PBX send to the gateway? There should be an INVITE going to the 127.0.0.1 IP address. WHat are the headers looking like?

In the beginning, we actually had the MoH file open always at the beginning. It sounded very strange, because people often get put on hold more than once during a conversation. That is why we introduced the seek into the MoH files! MoH for each CO-Line? I believe that could be solved by using a different domain for each business.

The point behind having the domain dropdown is for hosted environments where ther operator loads a MoH file specifically for a customer, so it should not be visible for other customers. On the CS410 is has limited use. If there is only one domain then there is no point in hiding it! Don't forget that you have to put the file on the file system. The only way to do this on the CS410 is through sftp.

Of course, you can choose a different name when writing the file to the local PC. I agree it would be better if the message carries some more information about when, who and so on. For now the workaround is to leave that decision to the end user.

You mean you want to hide the caller-ID for outbound calls?

Did you see the speaker symbol in the web interface (when you log in as user and check the mailbox)? There you can download the WAV. Downloading even marks the message as saved!

We asked the chip vendor for an update, but so far nothing. The biggest problem is the threading library, which cannot deal with threads in different priority queues. We currently have a workaround that could come from the good old Windows 3.1 times, but that eats a lot of CPU power. The other issue is that it would be nice to have a "IPv6Ready" certified kernel. The software is "ready", but the OS also needs to be ready.

"It depends". For a 10-user installation how have to deal with maybe 40 MB, for 100 users it can go into the 100-200 MB. The number of calls also does matter. Fortunately, memory is cheap these days and usually not a problem.

Smells like a DNS problem to me... Maybe you can use the IP address in the outbound proxy instead to see if that is the problem. Keep the DNS name in the domain, so that Exchange does not complain.

IMHO MPLS is perfect for running the PBX in hosted mode for small companies. I would like to know if there is any experience out there with this.

Can you try it out? We can make a test build for CentOS.

It looks like we have a race condition here. When a message is being recorded it takes some time (ms) until the file made it to the file system. If the check if the file is in the file system is quicker than the file system write, the PBX might really think that there files is not there and it will delete the message. Workaround: Hard to say. Probably we need to put in a extra safety belt that does not delete "young" messages.

I agree. Lets see how quick we can incorporate it.

Not sure, but you might need a pattern and replacement like this: Pattern: (\*[0-9]*)@.* Replacement: sip:\*\1@\r;user=phone A lot of baskslashes! IT world meeting telephony world...

Check if you can make an outbound call to the number from that phone. If that does not work, then click 2 dial will also not work...

That will not work. Multiple domains require either an outbound proxy in the SIP phone or proper DNS entries.

No, when you are listening 1 is rewind backwards. You have to press "6" to move or copy it. Then you get into a special menu. Make sure you have the latest WAV files; there are some new prompts; if you don't have them you'll hear digital silence (there will be a LOG message about this).