Vodia PBX
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Turn off silence suppression on your phone - the PBX probably thinks that this phone already died (silent as it is).
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If you like, try the attached WAV file as ringback file. There is also a busy file available.
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so paging through the phones is useless.
Well, that might be your conclusion.
Most phones actually do not put a call on hold when another call comes in.
If you have such a phone, open a trouble ticket with the phone vendor and/or set the number of lines for that extension to 1.
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This feature is very useful for us as we have the pbxnsip license controlling number of concurrent calls and we need to keep an eye on it.
Okay. BTW you can also use SNMP to track the number of calls: http://wiki.pbxnsip.com/index.php/SNMP.
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This feature is very useful for us as we have the pbxnsip license controlling number of concurrent calls and we need to keep an eye on it.
Okay. BTW you can also use SNMP to track the number of calls: http://wiki.pbxnsip.com/index.php/SNMP.
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Here my log while calling. I'm not sure if you can spot something unusual -
No really. I think Wireshark would be more useful, or a SIP-trace.
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Is version 2.2.0.2415 for SuSE 10 available?
No (on the head versions we usually have Windows executables).
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Still sounds a little bit like a mystery to me... It would be great to know what exactly is causing this, then it is probably a five-minute fix.
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Yea we took that out - it is still there in domain mode, but in admin mode we took it out.
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For that you need a "PSTN" gateway. It is not really PSTN that you are calling, but from the PBX perspective that gateway should behave like a FXS (so you need a FXO gateway).
Examples would include AudioCodes, Dialogic, Grandstream, Mediatrix, Patton, Vegastream, just to name a few great products.
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The 1.5 version also offers SRTP even when a insecure transport layer is being used. But there is a flag in the admin settings that says "offer secure calls" or so at the bottom of the admin settings, try turning it off. Then you should not see the SRTP keys in the SDP of the INVITE message any more.
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That would be very difficult - a SIP phone does not send a notify to the PBX when it goes offhook.
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Plus it is very simple to implement!
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Unfortunately it seems the new version for SuSE 10 is still unavailable.
Check out http://www.pbxnsip.com/software, http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2115 should be available!
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Eehhm... Multiple domains are a complex topic. If this is your first installation, I would recommend to use only one domain and get some experience first.
Also, check out http://wiki.pbxnsip.com/index.php/Log_Access for the logging issue. See www.wireshark.org for a tool that helps you find out whats going on.
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I think then the best is to take a look at a Wireshark trace or at least at the SIP trace of the involved user agents.
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Is that phone registed in the same LAN?
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In 2.1, there is a "night service" with the name #l (pound L) that acts like a service flag when all agents are logged out. See http://wiki.pbxnsip.com/index.php/Agent_Group#Night_Service.
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... but inter office calls from office A to office B here a static sound about 6-7 minutes into the call then it drops ...
That sounds like SRTP trouble. What versions are you using?
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No it seems to be fine on the PBX as far as we can tell... In the real world the problem is that the support from PSTN gateways is very poor for the RFC.
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Is it possible to have a string similar to $r/rec-$d-$t-$i-$u-$n.wav that will allow the Recordings to be saved each day in a seperate folder for that day on a different Hard Drive?
Did you try $r/recs/$d/rec-$t-$i-$u-$n.wav? If you link the recs directory to another hard drive, then the server will write it in the other location. In Linux that would be a symbolic link, not sure on Windows.
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VPN on a phone is a difficult topic... It is very useful if your SIP infrastructure does not support application layer security.
The good thing about pbxnsip is that TLS and SRTP already keep your voice pretty private. I think it is much easier to go this way.
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Looks like we can change the NTP strategy now. If Linux has a time, then there is no need to do big changes in the PBX any more. We will just check for time drifts as the internal clock is not very precise.
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Oh there is a change in the setting "Explicit Remote Party-ID" (which is now DID): http://wiki.pbxnsip.com/index.php/Outbound...Remote-Party-ID. Maybe you need to change that setting.
Linux Setup
in Linux-related Topics
Posted
I would recommend to get started with Windows first. Having a new product (pbx) and a new operating system (Linux) will make this project hard. Once you got the PBX under control in Windows, you can just copy the files and continue in Linux.
Our Linux build does not support dongles yet.