Vodia PBX

Are we talking about the latest and the greatest? http://forum.pbxnsip.com/index.php?showtopic=340 solved many RTP-related issues.

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.

Working on it...

Clarify: You mean when you listen to voicemail and then press '7' you have this effect? You hear that on the phone?

Are you using TLS? Are you able to see the SIP traffic? The PBX uses SRTP only if TLS has been used. Versions? Of the PBX and of the phones?

Well, the PBX code only deals with WAVEHDR, which contains a block of data. I would say this problem sounds like a problem with the audio subsystem as the PBX does not even dive into the byte area. Instead of just blaming the audio subsystem, would it be possible to have a similar setup (maybe different computer, same type) and see if the problem also occurs without the PBX being involved?

Nope, the file must be on the local file system now. Should not be a too hard requirement IMHO.

Well, return the device to where you got it from... Possiblty including a reference to what device it was, e.g. the MAC address.

Now you need to put a file path there, e.g. img/logo_custom.gif located in the html/img/logo_custom.gif file relative to the working directory.

There is something in the Wiki: http://wiki.pbxnsip.com/index.php/Processi...DR_from_the_PBX. It does not reflect the latest add on's in 2.1, but the core has not been changed.

See: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2108.exe http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2108 It seems that the RTP pass-through problems are fixed. Also, this version now always sets the processor affinity to a value (1 is default now). Looks like this version solves a lot of problems.

That sounds like the PBX is missing a byte in a 16-bit sample?

Well the Alert-Info header is controlled by a XML file (see below). Maybe it needs to be tweaked for the GS again. But most of the phones now understand this way of Alert-Info. If you change the file, you can now re-load it through the web interface (admin/config?) to that you don't have to restart the service. <?xml version="1.0"?> <ringtones> <tone name="custom1"> <vendor ua="Polycom.*" type="alert-info">Custom 1</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom2"> <vendor ua="Polycom.*" type="alert-info">Custom 2</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom3"> <vendor ua="Polycom.*" type="alert-info">Custom 3</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom4"> <vendor ua="Polycom.*" type="alert-info">Custom 4</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="internal" type="internal"> <vendor ua="Polycom.*" type="alert-info">Internal</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr2></vendor> </tone> <tone name="external" type="external"> <vendor ua="Polycom.*" type="alert-info">External</vendor> <vendor><http://127.0.0.1/Bellcore-dr3></vendor> </tone> <tone name="intercom" type="intercom"> <vendor ua="Polycom.*" type="alert-info">Auto Answer</vendor> <vendor ua="Cisco-CP79.*">auto-answer=0</vendor> <vendor type="call-info"><{from-uri}>;answer-after=0</vendor> </tone> </ringtones>

Well the paging stuff seems to depend on the phone type (and possible software version). What phones are you using?

Well, 2.0 had hand-crafted images for the menus on the top. That was obviously not very smart and has been replaced in 2.1 with a simple html interface that just uses a few background images (some people were asking if we can change the color and the answer was "ouch"). F5 is your friend now. If you use your own design, most of the logic should still work, but just make sure that the cor_xx.gif images and the main_*.gif stuff is moved away. Sorry, but in this case it is really hard to stay 100 % backward compatible...

On a scale between 0 and 10, how useful is http://wiki.pbxnsip.com/index.php/Audiocodes?

The clear log is fixed. *87 must always result in audio, if there is no pickup available then there should a IVR annoucement.

Today we made another version. There was a problem with the RTP pass through and the keeping of the SSRC and the packet numbers. Fingers crossed this topic can be closed now. Maybe it was also the reason why people were reporting DTMF problems. http://www.pbxnsip.com/download/pbxctrl-2.1.0.2107.exe

Version 2.1 has global settings called "timeout_hold" and "timeout_connected" which can be manually edited in the pbx.xml config file. You can set it to any value bigger than 10 seconds. Depending on the phone and the software version, the phone should send some kind of keep-alive traffic during mute. Silence does not mean there is no RTP!

Yes, http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2105.

Yea, we changed something to make logging faster (and avoid locking out other threads), and the clear log was overlooked. Should be fixed in the next. Thanks for reporting!

Try "sip:192.168.1.3:5061;transport=tls" as outbound proxy. If you see "Sent to udp:192.168.1.3:5061" then the phone uses the wrong transport layer. I guess you should also reboot the phone after that. And try to use version 7.1.19 or higher on the snom 370, previous versions might be buggy. See http://wiki.snom.com/Snom370/Firmware.

Sorry, should be there now. I think so. Outstanding issues (top of my head) are: (1) Some strange stuff with the mailbox when you enter the wrongpin the MB starts recording itself (?!). (2) CO-line monitoring seems to be fixed, but need to verify. (3) There is still a case where T.38 does not work (properly?)

Can you post a screenshot or something like that? Would be great if others have the same problem...

We are still testing and finding issues. So for production the latest and the greatest is still 2.0.3.1715. The latest Windows build of the day is 2.1.0.2105 (http://www.pbxnsip.com/download/pbxctrl-2.1.0.2105.exe), if you like you can try this one out and help exposing it more to the real world. We are running it already, but still have one or two tickets open that we want to close before releasing it. Once we are through with 2.1 QA, one thing is sure: This will be a great release addressing many features requests (TAPI, email spooling, G.722 and Re-INVITE, multiple mailbox messages, ...) and fixing several issues that we had in 2.0.3.