Vodia PBX
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The problem here is that obviously his phone completely stopped sending RTP and also stopped sending anything else. For the PBX, it has to make a decision: If that phone crashed, this call would still be billable.
Usually, a phone should send some silence indicators or at least some empty packets to indicate that it is still alive and there is no reason to disconnect the call.
Cell phones also send silence indicators. Whops, that's TDM - if you put the cell phone in Farraday's cage I guess the call will disconnect much faster than in VoIP.
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You can do that using a tel: alias for the extension that you want to use. See http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk.
For example: Set an alias for the auto attendant to tel:2121234567 and another tel:alias for the extension that you want to route to tel:2127654321, then you don't have to set anything in the extension field of the trunk.
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This can happen if there are more than one phones registered to the same account and one account does not subscribe to the MWI state. Are the registrations stable?
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There is a new version available that you can use. Everything about this image is in http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.0. The image can be downloaded from http://www.pbxnsip.com/download/pbxctrl-2.1.0.2084.exe (Manual Upgrade).
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That depends on the phone and there are several ways to do it.
One way it to use a DNS SRV entry to locate the PBX. In this case you need to make sure that the DNS name of the domain (probably then not "localhost") exists also as a DNS SRV name.
But probably it is much easier to tell the phones to fail over to a second registration of the primary registration should become unavailable.
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1) Can keep the extensions we currently have for the hard extensions: 1XX
Yes, but you need to put the MAC address of the phone into the extensions setting (registration tab).
2) From what I read to use PNP you must use TFTP, is there going to be a change to have FTP/HTTP instead?The PBX does not support FTP, and at least in the old firmware versions the only other option was TFTP. But the first small TFTP file does nothing else than redirecting to HTTP. If you are able to direct the phones directly to HTTP, then you should be able to skip TFTP completely.
3) Can the system dialplan be edited, I don't like the generic ones and I want users to dial with a 9, etc so that we can transform as needed and not interfere with extensionsYes, there is a seperate file called "diaplans.xml" (I attach the most recent version) which you can edit. If you put that into the html directory of the PBX, it should pick it up during a restart of the service. Don't forget to move this file away if you later make an upgrade and you want to use the default method again.
4) When provisioning the defualt config, can that be edited to allow for failover, etc?Yes and no. Firstly, we want to avoid supporting a very broad way of configuring devices to keep the support effort low (that's the "no"). But you can always put other files into the html directory to override the default. You would need to get those files from us, I don't want to post them here.
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That is typically a SRTP problem. If you don't need SRTP, just turn it off. Otherwise you must match right versions...
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Polycom phones support the presentation of dialog state. If you enter a list of accounts (e.g. CO-lines, extensions) in the extension settings ("watch calls"), the PBX should automatically generate the right settings for the phone.
Alternatively, you must provision the phones manually. I think it was the address book that needs to list the "buddies" that you would like to see in the screen.
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Hmm.
Are you using an external voicemail system (Exchange)?
Is there an overlap with another account? Or with a direct destination? Any log messages?
When dialling a mailbox the auto attendant will check if there is a match, the settings on how many digits it should look or is not being used. As soon as there is a match it will redirect the call to the mailbox.
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There is a setting in the domain level for the 8 prefix. Maybe you can change that to 88 or to another convenient combination to tell the PBX how the users can reach the mailbox directly.
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The way to go is:
When the PBX starts up, call ntpdate to get the PBX into a reasonable time zone. For this step, ntpdate must be on the box and the startup script must be adjusted.
Then the PBX will perform a internal NTP date lookup every hour to adjust the differences to the hardware clock in the PBX. For this step we must make sure that the setting ntp_server in the PBX is set properly.We will include that in the next software patch for the appliance.
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The PBX runs in one process. That means if the host has multiple CPU, the operating system can distribute several processes over the different CPU, but it can not split up the PBX process over CPUs. Practically that means, if you run the PBX on a machine with two CPU, the OS can put other tasks like file system and networking on one CPU and the PBX on the other CPU. Having more CPU will likely not increase the performance of the PBX.
Hyperthreading means to me that there is one CPU, but it can physically handle several threads at the same time. Because the PBX uses multithreading, that gives a real performance advantage over a non-hyperthreading CPU. However, there is also a limit. Because the media processing is done in one thread, that thread is in the critical path.
I would say, if you want to have a good performance price ratio, go for a hyperthreaded CPU with either a single or two cores. Better spend the money on a high-performance I/O subsystem and double power supply, so that that machine will not cause costs by hardware failures.
1 GB RAM should be enough.
If you have abundant CPU resources, you can run several PBX on the same host. You can do that either by running them on different ports (maybe using DNS SRV to locate them). Using different IP addresses might be a challenge for the routing subsystem, so I would stay away from that at this point.
And I would be careful with virtual machines, IMHO they are not very well tested with real-time requirements such as RTP handling.
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The http_rate is a setting that protects the web server. If simply says how many HTTP connections are accepted per second. The default is 5. Usually the web browsers re-use one connection, so that 5 per second should be a reasonable value. If you want to use SOAP and open a new connection for every request, you might need to change that value.
There is another setting called "max_udp_invite". This setting limits the number of new INVITE requests coming in over UDP, which effectively present call attempts. The default value is 10, limiting the number of new calls per second to 10.
INVITE over TCP or TLS are handled differently. Because the PBX needs to accept a connection, we have a TCP attack problem anyway. The PBX simply waits 500 ms after every TCP connection. That might lead to a waiting line for fresh registrations, but for registrations waiting a few seconds is usually no problem. Then the number of SIP DoS attack over TCP/TLS is not as severe as over UDP.
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Check out the latest Cisco financials and how much they made on VoIP. They were smart enough to come up with a solution that is pure IP, and they got it working. Doing the same thing in SIP instead of Skinny is more complex, but the result will be even better than a closed solution from one vendor. Then check out the latest sales report about the WiFi-based phones. Maybe Cisco buys Nokia to keep their solution closed!
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Loading a certificate into the PBX is quite easy. The PBX does not really parse the content, just presents it during the SSL/TLS setup. Then it is up to the client to trust it or not. So I think it should work and is worth a try.
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Hmmm, I was more thinking about accessing one existing account and service... If you can share login information, please send a private message to me or to support@pbxnsip.com.
Can you confirm that PBXnSIP is supporting RFC 3407?No. But it does not have to be compatible (that's the purpose of RFC 3407). The PBX does not need to have capabilities outside of the "classical" codec set. The negotiation should work fine, there must be something else trivial in the way.
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The SDP from the Nortel looks a little bit unusual, but not wrong. Is there any way to test against this device from a public IP address?
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Ouch. That is exactly what we wanted to avoid...
I am only thinking about a curl script that logs on, and then presses the reboot button. If the web interface is still alive, then that could be a short-term workaround.
Actually, we are working on a watchdog timer as well, but it is not ready yet.
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What about just using "account@domain.com"? We are using this and that is a workaround until the string presentation in HTML is fixed.
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Yea in 2.0.3.1715 it is still a little bit "picky" and the example actually does not work. There was something with the quoting/escaping in the html forms I think. Try something like PBX <pbx@domain> or just pbx@domain.
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If you can dial into your extension's mailbox from the outside: Press *98123# (an extra pound at the end). This is like calling from your extension the *98123.
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If you are connecting two PBX avoid auto attendants. You can usually directly dial into a specific extension and there is no need to two-stage dialling.
In the latest version we added the possiblity to keep the original From-/To-headers so that it is easier to keep the routing information when using several PBX linked together. But we still don't have a good stable release with this.
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If you like, please try http://www.pbxnsip.com/cs410/update-2084.tgz. This tarball includes the ntpclient and also makes sure that it is called after startup. Other things it does is AGC on the FXO side, which should improve the audio quality and reduce problems with echo.
To load that image, just follow the steps in http://wiki.pbxnsip.com/index.php/Applianc...the_PBX_version.