Vodia PBX
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I would kill 269 (bash with pbxnsip start)
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In fact . . . I just did a test . . . I turned on the unit and started pinging the IP address that the DHCP client was handing off at the same time. You can see exactly how long the unit comes online and then subsequently fails. Here's the interesting details:
Hmm. That more looks like a problem with the startup - maybe file system full or something going wrong with the firmware update? Maybe a problem with the watchdog timer? In any case, it is worth to try to SSH in and use the few seconds alive to kill the last bash process. That should hopefully stop the madness and give you time to look around what the problem is.
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at System.Int32.Parse(String s)
Okay, not beautiful, but it seems that you have an Extension name which is not numeric... Don't ask me why, but that is what is in the code! Can you pick a Extension that is a number? [*blush*]
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Oh it seems you are using an old version of the PAC. This version had a bug with the parsing of the responses, and was not able to safely locate the attachments (sic!). Try this link: http://www.pbxnsip.com/pac/setup.exe, that should get you the latest and greatest.
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I'm using 2.1.6.2450 - and I could not duplicate the issue fixed in 2.1.8.
The PBX could get into en endless (deadly) loop in 2.1.6, that's why we recommend to disable the camp on in this build. 2.1.8 fixes it.
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No, there is no redirects. Just IVR node with greeting, no records in CDR. It happens when I use "Send call to extension:" option only.
Very strange. We need to keep this on the radar...
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This is a long long discussion. IMHO everyone should follow RFC3325 and we are all set.
Until then, all you can do is playing around with the different methods in the trunk and see what works best. But many ITSP today do not support sending redirected caller-ID (e.g. for a redirected call from an incoming trunk to the cell phone).
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I have found that on a Snom 360 if there are different buttons programmed to be different Park+Orbit destinations, the park reminder feature does not work. however if you manually put the call on hold, and dial *85+orbit number then the reminder feature works. Does anyone know how to fix this?
No, don't use that. The parking there has not much to do with the parking on the PBX. Better use the buttons package, there it should be working fine (see http://wiki.pbxnsip.com/index.php/Assigning_Buttons).
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Hmm. We can send either Alert-Info or Call-Info header, I don't recall how the Cisco 7960 likes it. BTW what version are you running?
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We have configured Trunk for incoming to PBXnSIP calls. This trunk configured with options "Send call to extension: 111" and "SIP Gateway". When call arrives from this Trunk - PBXnSIP connect the call with IVR Node 111. Working well.
But I don't see this calls in Status -> Call Log..
Also nothing in CDR...
Is that call being redirected somewhere? Does the final transferred call show up in the CDR?
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I am also trying to use IVR node. The same, IVR node repeat the prompt and repeat when user don't enter the digit. How to disconnect the call or redirect this call to another extension when user do not pressed any digit? Have you simple example?
You can handle timeouts in IVR nodes: http://wiki.pbxnsip.com/index.php/IVR_Node (just updated).
So - just enter a timeout value and use the pattern "!T!123!" in the DTMF Match List if you want to redirect the call to 123.
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Hmm. Progress, but still I don't get it 100 %.
When the PBX performs the T.38 switch over, it must choose another set of ports. That is because T.38 is "image", not "audio". That has also the consequence, that the ITSP must learn a new NAT binding for the T.38 ports.
When the PBX learns the T.38 destination, it should write something like "Passthrough: Changing destination to ..." (Media, log level 8). Do you see something like that in the log?
One little problem with T.38 is that there is no RTP header. That means the PBX will turn it's head around for any kind of junk hitting the port (if the transmission stalls for more that 100 ms). But I don't think that this is the problem here.
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Did you set up a button group for the extension that your register with?
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Can Metropolis show information like waiting time and that this call was a ACD call? How should it look like?
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The bind() message should not be the problem (we already took it out in a later build). The PBX just tries to get a port, and if that port is not available, it tries another one.
Of course, make sure that you have a large RTP port range, so that you are not running out of RTP ports.
I still don't clearly understand what makes the difference between the failed call and the successful call. Do you say that is depends on what port it chooses? If that is the case, can you double-check the firewall port range?
Other potential reasons for such behavior are usually race conditions, e.g. the answer comes earlier or later. Is there anything in this direction?
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It does not follow any pattern; it has happened on cell phones, land lines, phones with hard-wired handsets, speakerphones, cordless phones, anything, from as best I can tell.
Hmm. Maybe you can call the mailbox or some other IVR which has "digital" audio quality (e.g. the conference room without anyone in it). If the problem persists, then maybe there is some parameter on the conference phone wrong? Maybe the echo of the room generates a loop or something like that.
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Off topic slightly - How often are the Admin status reports (the graph) - scheduled to send out? I seem to get them every other day or every 3rd day.
You really should get them every day. We do get them every day (and it has become a valuable morning lecture for me). In the beginning they were often classified as SPAM, but once that we told the email system it is not SPAM it works quiet stable.
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Well, there are two things.
First, there are two places for settings the SMTP server. One is on admin level, and there is another one on domain level.
The other thing is that once that emails are being put into the spool directory, the email server is fixed. That means, you might have to manually delete the emails from the spool directory to get them out of the system.
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Yes, it is on the short list. But it will be in the version 3.
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It would probably better to split this topic off to a new topic, as it is not directly related to the 2.1.7 release.
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how cool would this be to have a binary version for mac os x?
this should just be a recompile with xcode no?
There are a couple of differences. MAC OS is a little bit like BSD. Not sure if MAC is a significant market, we were also thinking about Solaris. Sun has a very good reputation in the server market, and that is what you want to hear when you are talking about a PBX platform.
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It would be nice if it would also work with pidging/adium, they are just chat clients and thus only support SIP/SIMPLE and not a full SIP stack.
Do you have a LOG with the SIP messages? No problem if they don't support other messages, they only need to support REGISTER, MESSAGE and properly reject incoming INVITE messages (e.g. with 500 Not Implemented).
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The SIMPLE support is very simple. The PBX practically works like a SIP proxy in this case, it just takes a MESSAGE request and sends it to all registered user agents. There is no store/forward. I would say, register the device and give it a try.
So far I have seen it working with Counterpath (did not try anything else).
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When I hear "static" the first thing that comes to my mind is SRTP trouble. Is there a crypro heder in the SDP? If you can still hear the real audio it is not SRTP.
The PBX itself is "pure digital" and does not introduce noise into the RTP stream. Does it make a difference what you are calling? Maybe there is a audio loop if you are talking to something in the room or in the office.
Configuring Polycom SoundPoint IP 430 w/pbxnsip
in Polycom Phones
Posted
Did you drop the firmware files in the tftp directory of the PBX? Also, is there at least one extension with a * in the MAC address (in the registrations tab)?
Another common pitfall is that the PBX does not (by default) provision the password. Check the admin/settings/ports section on the password policy.
You can check what has been generated in the directory "generated". There you should also be able to see if the password has been provisioned.
Some more information at http://wiki.pbxnsip.com/index.php/Polycom.