Vodia PBX
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2.1.8 is available. The only thing changed was a neccessary fix in the camp on feature. Workaround is to disable camp on or upgrade to 2.1.8. See the release notes at http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.8.
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No. Moscow time is GMT+3 with DST.
The question is if the DST dates are the same like in CET. Of course, the offset from GMT is different.
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Did you apply that latest MSP tgz?
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SNMP: Well we could add the standard sensors that run on a PC. The box itself is just a plain Debian 3.1 computer, and it should be possible to get the standard SNMP stuff working as well.
CPU is a problem, there we are depending on the chip vendor.
RAM should not be a issue, 128 MB should be more than enough for a task like this.
Flash: Next batch will have 1 GB flash, should be also more than enough for most installations...
What we want to include is a DHCP server that provisions option 66, so that pratically all SIP phones can just plug and play.
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So what was the trick on the PBX? Turning off UPDATE? Or just configuring the HT286?
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Has there been any progress on the Linux/dongle effort?
No this is still in limbo. Unfortunately, but it is still on the list.
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Can you exclude a hardware defect? Does the web interface in the PBX show that the phone is registered?
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I don't know when it happens... I'll try and reproduce... Do i need to log in as a user?
I think we could reproduce it already - it was the status page.
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We have that combination in one of our closely monitored installation, it works if you turn support_update off (set it to false).
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[1] 2008/04/09 20:00:09: Unknown SSI command dbh
[1] 2008/04/09 20:44:11: Last message repeated 140 times
Ouch! Does that happen when the user goes to usr_status.htm? Seems there is a leftover from previous versions.
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Yea, this area is a little bit buggy... There is also the problem that when the user hits pound after recording, then the message is treated as already in the inbox. We need to perform some cleanup in that area.
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Maybe it would make sense to have an upper limit on the DNS expiry time. Many DNS records have a duration of a day or so, which is very very long.
Alternatively, maybe we should add a button that just clears the DNS cache. That would also force a reload on the next occasion.
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I am using Polycom's; therefore no Record Button.
Is it possible to just send the DTMF to the PBXnSIP FS, and not to the end user? Especially when it's preceded by a * code ?
I was thinking if we could use one of the A-D DTMF codes, which are not very known. Then we could block the relay. But I have no idea on how to program a key on the Polycom so that it sends out such a code...
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I am having the same issue. Customers coming from traditional PBX systems expect to be able to monitor, and speed dial the extension all on the same button. I feel that not having this feature is a major shortcoming, and leads to many unhappy customers. I did find that if you are using a snom 220 you can set the button type to "destination", and then enter the extension number in the format of <sip:204@yourdomain.com;user=phone;intercom=true> and the button will monitor, speed dial, and pickup ringing calls all in one. No one ever complains about accidentally picking up a call. I wish there was a way to get a snom 360 to do this. The snom 360 does not have the ability to set the button type to "destination"
If you select the "speed dial" option (in the PBX) with the buttons, it should work. Pressing the button always dials the same destination, no matter what the LED is showing.
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Very strange... Does it "heal" after some time??? The forcing of the re-register does not change the internal outbound proxy. But hitting the save button on the trunk should do, if something changes in one of the settings.
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One common problem with the Gradstream devices is that they advertize UPDATE, but when they receive a UPDATE message with a SDP offer, they don't answer this.
What you can do is to turn off the support for UPDATE. The setting is called "support_update" and is in the http://wiki.pbxnsip.com/index.php/Global_Configuration_File. I would try to set it to "false".
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That is probably because the firewall is SIP-aware. Sometimes the firewall wants to help, but just makes things worse. You can bypass the firewall using TLS (well, if that's okay with your security guidelines). Or just turn that part of the firewall off (actually, in this case the help is not needed).
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Hunt groups cannot respect redirect settings. Otherwise you'll end up very very very quickly in endless redirects. Plus nobody knows why the secretary suddenly has to answer the calls to the support hunt group.
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Well, how would the phone interact otherwise? You could try INFO messages, but then the pass-through of DTMF to external applications will suffer. The Record button solves that problem, if available (AFAIK only on snom).
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No, I'm referring to Dial Plan Entries, not Trunk settings.
Oh. In theory they should be the same. In 2.1 there is more logging, but that should not have any side effects.
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Now we have "follow what has been set in the list of agents" and least recent. Ringall can be done by increasing the number of agents called in one cycle.
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It is already fixed in head, possibly we can port it back into 2.1.8.
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That is a tricky topic. Of course the intention was to keep the CS410 on the same path as all other software - but we simply have the problem that we need the settings for CPC duration, polarity change and busy detection which does not exist in 2.1. There are no "dangerous" changes in the 3.0 version yet, IMHO so far is it safe to use it. The release notes include everything between 2.1.6 and 2.1.7.
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I think if the device is not registered it does not consider redirect after timeout. Then only the "redirect all" applies. Not very satisfying... But redirect after timeout is also not the answer. Maybe redirect on busy would be a solution in this case.
Frozen fxo port & email strangness
in Embedded
Posted
$ host smtp.mail.thesoundshop-llc.com
Host smtp.mail.thesoundshop-llc.com not found: 3(NXDOMAIN)