Vodia PBX
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2.1.0.2111 is out: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2111.exe, only Windows so far.
It fixes problems with certain versions of Polycom phones putting callers on hold (using a=inactive) and another Problem with Linksys phones that believe that codec 18 is G729a.
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Do yo usee SUBSCRIBE with the Event x-tapi showing up on the PBX? If not there must be something with the TSP setup. And you have to reboot your PC after installing the TSP and setting it up.
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We will later provide a .tgz that you can put into the web interface. The above is just the "naked" image that you would have to put directly into the file system by SFTP or some other mechanism (ftp).
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Hmm. Probably a problem with the case. The TZ parameters must be lower case, better keep the whole parameters lower case.
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The web interface feature will make it into the 2.1 release.
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Don't use FTP. Better use TFTP or (if you have PBX 2.1 or higher) HTTP. The PBX has a built-on TFTP server than can also perform software updates and on-the-fly generation of the configuration data for Polycom phones. See http://wiki.pbxnsip.com/index.php/Polycom.
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The 4-digit dial plans should be also there in the latest lang_xx files, so it should be okay to move them out of the way.
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That is really weired. I could hear the DTMF in the audio stream, it did not sound too bad.
Of course, jitter in the DTMF tones is very bad for the detection (that is why they invented out of band!), but I guess that does not seem to be the problem here.
[shrugging shoulders]
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Hmm. Can you get a Wireshark trace and post me a private message where I can download it?
Okay, got the PCAP. Did you turn inband detection on on the PBX? See the admin settings at the bottom. See http://wiki.pbxnsip.com/index.php/Overall_...gs#SIP_Settings. My success in detecting inband depends on that flag.
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If you only have those files there should be no problem. These files have nothing to do with the appearance of the web interface.
Apart from the browser's cache. Usually pressing F5 should solve that problem, maybe you need to explicitly delete the cache.
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Well, I remember that was also a point that I simply did not understand. Seems there is a code that needs to be cracked with those phones.
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Hmmmmmmmm.
Well, the only thing that I can think about right now it to use the latest and the greatest, which is currently http://www.pbxnsip.com/download/pbxctrl-2.1.0.2109.exe (see http://wiki.pbxnsip.com/index.php/Installi...#Manual_Upgrade on how to move to that version).
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Well, I can think about the following solutions:
1. Make a dial plan for each extension that routes the call to the remote emergency office (simple, but might be a lot of work to set up)
2. Use a user parameter that contains the emergency number. You can reference them with \x, \y and \z for the parameter 1, 2 and 3 in the replacement pattern in the dial plan (e.g. sip:\x@\r;user=phone). The pattern would be a simple "911".
I think 2 would be the most pragmatic solution here, but you must make sure that all users have the parameter 1, 2 or 3 set.
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Well, I guess we should at least add it to the web interface.
Having a star code would require that there is a permission concept behind it (not everyone should be able to change someone else's agent status). That would stop me from having a star code extension like *64123.
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You can use the star code mentioned in http://wiki.pbxnsip.com/index.php/Feature_...ogin_and_Logout.
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Inband detection fails. Transcoding works.
Hmm. Can you get a Wireshark trace and post me a private message where I can download it?
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Yea if you are moving to 2.1, check the html directory... If you really need to modify content, it is now time to merge in changes.
Before that, make a backup...
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Are you sure this is a problem of the PBX? Usually the SIP phone generates the DTMF tones for user input.
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Check out http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk.
When you are using the SIP Gateway mode, it is crucial that you are using the outbound proxy. The PBX needs to identify the trunk by the IP address, and for this purpose it uses the outbound proxy. Outbound proxy = inbound proxy.
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Okay.
I still don't understand why G.729 would improve that situation... DTMF was not featured in this codec (it must be out of band then).
Are we talking about DTMF detection or DTMF transcoding here? Detection should be okay or better than previois versions, transcoding requires that the PBX falls back to RTP disassembly, which it should do automatically (there should be a log message about that).
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Are we talking about the latest and the greatest? http://forum.pbxnsip.com/index.php?showtopic=340 solved many RTP-related issues.
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Inband: If you don't have to, then don't use inband. Especially on the embedded system, it cost a lot of performance to analyze the audio streams.
G729 pass through. Well, it might be possible to pass G729 through, but the big problem is what happens if someone hits the hold button or parks the call. The PBX then is not able to render audio... Therefore, G729 is still still a problem.
G722 is possible there, but you must edit the pbx.xml file and add the codec to the preference list (codec number is 9). We did some tests with the codec, tests show that it works-however it really has limited value as the preferred PSTN termination is FXO and that is quite the opposite of wideband audio.
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Working on it...
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Clarify: You mean when you listen to voicemail and then press '7' you have this effect? You hear that on the phone?
Okay, 2.1.0.2108 is out
in New Features/Versions
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G722: Yes. Also known as codec 9. But you must list it in the codec_preference setting.
Image: I attached a screenshot of something that works.