Vodia PBX
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Posts posted by Vodia PBX
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Is it done yet?
In head, yes. Actually no third mode, just filtering for edge state changes. That should do the job without too many changes in the code and in the web interface.
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Should I change the pbxnsip to 0 only?
I would give that a try for the sake of finding out if that is the problem. Then we can decide what to do with it.
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I did set the default codec to g711u on pirelli end to g711u. Can it have anything to do with alert info Ring tones, the only other thing I did other than upgrade was played with the custom4 ringtone in the ringtones.xml file.
Did you set it to use only g711u? Currently it answeres with multiple codecs, and that is a common source for problems.
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We have been to a couple if SipIt since SipIt 8 in Cardiff (different company at that time). Fortunately the SIP interop problems that we have in our day-to-day life are not the biggest problems (thanks to the B2BUA nature). You can't test a hunt group there, it mostly about spirals, tags, etc. It would be exciting if companies like Microsoft or Cisco show up with their flagship products.
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Maybe we can introduce a third mode for the flag in addition to "manual" and "automatic", which would be "semi-automatic" - this way we can definitevely keep backward compatibility.
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Yes, i read it.
When I call to my pbxnsip it's working normally, but when I call out from my pbxnsip it's disconnectiong after 23 or 24 seconds, the pbxnsip is directly connected with the ADSL modem, with firwall turned off.
Changing the registration keep alive doesn't help.
Do you have the SIP packets that cause the disconnect? Turn SIP packet logging on ("other" packets). If you see a CANCEL, the PBX causes the hangup; if you see a 200 Ok as a response to the INVITE, then the call was really connected. It sounds like the call was never connected. Maybe a problem with the provider, a SIP trace will reveal more information.
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Exchange really asks the PBX to call "5001;phone-context=PBXnSIP-exchangesp1.ourdomain.com", no kidding (no idea why the number must include all those strange parameters, but it is legal according to the RFC). The dial plan now must match also those strange parameters!
My idea for ERE in the dial plan:
([0-9]*);phone-context=.*
That should really fish out only requests that were initiated by the Exchange. Maybe a feature!
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RTP-TxStat: Dur=6808121,Pkt=0,Oct=0
Ouch.
Hmm. I think I am using the same device from the same manufacturer, no problems...
Can you set the codecs on the Pirelli or the PBX to use only one codec? Maybe that helps to track the problem down...
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It is the phone. It says it needs to be dialed using the e164 dialing or something along those lines when you dial without the + sign.
It is a new installation.
I don't know. I have never seen a installation where people had to put a + sign at the beginning of a number in order to dial out. I would rather look at the dial plan and then maybe put the + at the front of the number.
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I'm using a registered version. What can I do to solve this?
Did you already check the Wiki? http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems
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Well you still have the option to configure the device manually.
And apart from that I would put the files that the phone downloads directly into the tftp directory. According to the above procedure that should send the neccessary information directly to the phone.
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Whow that seems to be a common problem these days... Did you see http://forum.pbxnsip.com/index.php?showtopic=714 ?
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The problem with 80xx is that it would put the whole xxxx number into the replacement.
\1@.city1.domain.com seems to be buggy, there is a dot after the @ that seems to be incorrect to me.
If you have only one domain on a PBX (which is good!) then I would always use the name localhost, at least in the alias list of names.
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Well, first of all, set the log level to 9 and then you see what ERE the system is generating from the dial plan. That makes things usually much easier.
When doing ERE, please remember that the pbx must match a string like user@domain, not just the user part. That means a ERE like "50([0-9]{2})" is not complete, you need something like "50([0-9]{2})@.*". Using 50xx might be a problem because then you will reference the whole match in the replacement. If you can use 50* (variable length) then things are simple.
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When the extension has no registrations, if the service flag is Clear, incoming calls to the extension are answered by the voice mail. If the service flag is Set, incoming calls to the extension are redirected to the cell phone number.
You mean you expect the reverse logic? The service flag should have a function like "office hours", that means you specify when the cell should to be included in the call. That means if you say e.g. 9AM-6PM then you will hear the cell phone ringing.
IMHO most people would like to specify when they are available on the cell phone. Our first experience was that we got calls at 3 AM in the night on the cell phone, and it became clear that we need to specify the operation hours when redirection should be allowed.
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Is 2.1.6 - latest version being run in a production environment by anyone? And how long do I need to wait to see if problems are going to occur... I'm trying to quit using my customer base as a testing ground for new releases.
2.0.3.1715 was a great release.
Yes, we are running 2.1.6.2448 in several locations in real life. For us it works, of course. But that is not the problem.
We created a 2.2 branch for new features. 2.1 will be only used for fixes to make sure that we have a sock-solid build again. We did run millions of calls through the 2.1 branch, but the environment is not changing much during the tests. The real life is different, and we will see in the next couple of days and weeks if there is anything else that we need to address.
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I would then switch to SuSE10. The Wiki should point you to the most important things, but it does not replace a substantial know-how about Linux and SuSE.
If you are not familiar with Linux, then consider moving to Windows Web Edition. It is surprisingly affordable!
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[5] 2008/02/17 11:55:05: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT
You mean after hanging up the PBX shows that there is a call going on? Is that in the light on the CS410 (front panel) or on the phone? If the call is disconnected, the PSTN should just remain in idle state. Maybe there is something strange happening after hangup, maybe the 2400 sends some additional information after hanging up?
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2448 fixes a license problem that occurred with the 2446 version. 2448 is available as tgz that you can apply from the web interface on the cs410. So I would say this is the best and easiest version for the cs410.
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We were thinking if we should start another project that plays MP3 files and convers them to linear, then streams then via RTP to a local MoH port. But maybe it makes sense to search if such a project already exists and just properly document how to use it. Unfortunately, MP3 players seem to assume they run for a few hours maximum, so they don't have to take care about memory leaks.
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Dirty workaround... Lets take it as temporary solution.
Maybe later we come up with a nice idea that does solve this problem automatically - without side effects that we don't want.
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How does the PBX evaluate those signals? I have inband DTMF detection turned off in both systems, so it must get it via a SIP message? I dont know how that works but isnt that just one message saying the caller pressed 1 and not continously sending that tone?
Well if the PBX receives a RFC2833 packet and is not already in a RFC2833 stream, well then it assumes the key has been pressed. It does not care that another system already chopped off the first part.
So with the IVR node I would first send the call from the auto attendant to a local IVR node and that one would transfer the call to the other system after playing a blank 1 sec wav ?No, the call should first go the the IVR node and just don't do anything (no not react to DTMF at all). Then after 1 second or so the call gets redirected into the "real" auto attendant.
Phone requires + to dial out
in Dial Plan Setup
Posted
Maybe you should try a dial plan with the pattern 011* and the replacement +*... Then users can call 01133123456 and the ITSP will see +33123456.