Vodia PBX
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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?
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Nope, the file must be on the local file system now. Should not be a too hard requirement IMHO.
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Well, return the device to where you got it from... Possiblty including a reference to what device it was, e.g. the MAC address.
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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.
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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.
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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.
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That sounds like the PBX is missing a byte in a 16-bit sample?
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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>
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Well the paging stuff seems to depend on the phone type (and possible software version). What phones are you using?
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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...
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On a scale between 0 and 10, how useful is http://wiki.pbxnsip.com/index.php/Audiocodes?
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Are the log bits fixed in this? Also customer with *87 issues reported they've picked up a calls and got silence... Whether that is because they were too slow (or should they receive a 404?)
The clear log is fixed. *87 must always result in audio, if there is no pickup available then there should a IVR annoucement.
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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.
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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!
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Faulting application pbxctrl.exe, version 0.0.0.0, faulting module pbxctrl.exe, version 0.0.0.0, fault address 0x0016c030.
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!
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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.
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The last Windows build available on download is 2.1.0.2103 - could you please post 2.1.0.2105 for testing? Thanks!
Sorry, should be there now.
Does 2105 fix the ACK issue for paging?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?)
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Can you post a screenshot or something like that? Would be great if others have the same problem...
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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.
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Look at http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems for trunk problems.
Also, you can look at the BYE packets send from the PBX. There you can see how many packets the PBX received. In one-way audio cases, that is very useful. If it all does not help, you can use Ethereal to see what is "truely" going on on the cable level.
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There are images available for the Teles box that run 2.x. The upgrade should be relatively easy, once you are ablt to figure out how to put a new image there.
My suggestion is to wait until 2.1 is out and then upgrade to that version (we will include the NetBSD build for 2.1 as well). Should not be too long until then.
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What did you use for outbound proxy? Something like "sip:192.168.1.2:5061;transport=tls"? The transport=tls is important because otherwide the phone will use UDP.
Do you have a SIP trace from the phone's web interface? PCAP are useless, due to the nature of TLS we will not be able to see too much...
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Nono. The port number is not the point here. Proxy Server/Registrar Server are also not the point here. There is a setting called "outbound proxy" - I think it is in a different config page. In the config file it looks like this:
sip outbound proxy: {ip-adr}
sip outbound proxy port: {sip-udp-port}
No need for DNS. Lets keep things simple!
Then you should be able to put domain names into the proxy and registrar - they should be the same (e.g. company.com).
Snom and No Encryption
in snom Phones
Posted
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?