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Posts posted by Vodia PBX
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Cisco IS extremelly responsive and efficient with this product line.
Whow, I wish they had the same attitude with the 79xx line...
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I assume the problem is that SIP phones cannot ring forever. Usually the phone stops ringing after a minute. The PBX also has a maximum duration where it lets a UA ring.
You'll see that when you look at the SIP trace. Either the phone sends a code like "487 Terminated" or the PBX sends a CANCEL to the phone.
Having a phone ring "forever" is not a good solution. Imagine you are the caller and your destination lets you hear ringback tone for two minutes. I believe 99.9 % of the callers would already hang up after 30 seconds.
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Cheers, and still struggling to get *67 out the gateway..
What does the PBX send to the gateway? There should be an INVITE going to the 127.0.0.1 IP address. WHat are the headers looking like?
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On the subject of expanding the feature: MOH on CS-410
When the first caller is placed and remains on Hold they hear the MOH Music/Announcement from the beginning.
During that time additional callers placed on Hold will drop into the "Loop" and will start to hear what ever is playing at that instant.
Virtually all telephone systems use the "Loop" method for Music/Announcements On Hold but PBXNSIP is unique by starting from the beginning. Well, kind of.
This is OK, but businesses today want to fully exploit this captive audience area (Caller On Hold) and deliver a structured and precise message.
A caller placed in a "Loop" announcement may get the tail end of the message and miss the real impact of the main greeting.
Now, is it possible to have each caller that is placed on Hold receive the MOH from the beginning?
This would certainly set PBXNSIP apart from the crowd and be virtually impossible for traditional TDM systems to duplicate.
Lets push it even further:
How about a different MOH file for each CO Line?
Customers often times have a "second" business running in the same office.
This would allow each CO Line to play the correct MOH message to callers.
There is such an add-on item available, but it is unreliable at best.
In the beginning, we actually had the MoH file open always at the beginning. It sounded very strange, because people often get put on hold more than once during a conversation. That is why we introduced the seek into the MoH files!
MoH for each CO-Line? I believe that could be solved by using a different domain for each business.
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I have a customer with CS-410 3.2.0.3143 (Linux)
If I go to main administrative section -> MOH ,
I can create a file other than moh.wav . say xmas.wav... it appears that you should be able to have 3 or 4 choices here ,
but instead , it only saves the last option when i go to localhost -> MOH dropdown
I read the WIKI and it's not clear ,
is the intent that you get (1) additional choice for MOH per Domain , or should I be able to upload a few files i.e. Xmas , halloween etc for retail stores and let them change the music on hold to reflect the current season .. this seems to be the better option ...
so in a nutshell , is this a feature that we could expend on and make it better , or did we already make it better and a bug in there is only allowing 1 additional file at a time ,
The point behind having the domain dropdown is for hosted environments where ther operator loads a MoH file specifically for a customer, so it should not be visible for other customers. On the CS410 is has limited use. If there is only one domain then there is no point in hiding it!
Don't forget that you have to put the file on the file system. The only way to do this on the CS410 is through sftp.
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When accessing the personal EXTENSION mailbox the default SAVE-AS for recordings "MESSAGE.WAV" needs some more flexibility. Perhaps selecting a DEFAULT save option in settings page. Consider Perhaps allowing INBOUND CALLER ID and/or CALLED NUMBER
Of course, you can choose a different name when writing the file to the local PC. I agree it would be better if the message carries some more information about when, who and so on. For now the workaround is to leave that decision to the end user.
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No Love Moving to Dial Plan
You mean you want to hide the caller-ID for outbound calls?
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No love - Moving to FEATURE request
Did you see the speaker symbol in the web interface (when you log in as user and check the mailbox)? There you can download the WAV. Downloading even marks the message as saved!
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Are any plans in place to allow the embedded OS to get various linux updates? The Apt-Get configuration is configured for a private Debian Mirror. It would also be nice to load a few more diagnostic tools such as IPTRAF, IPFM and perhaps some OS SNMP tools for DISK SPACE or CPU.
We asked the chip vendor for an update, but so far nothing. The biggest problem is the threading library, which cannot deal with threads in different priority queues. We currently have a workaround that could come from the good old Windows 3.1 times, but that eats a lot of CPU power. The other issue is that it would be nice to have a "IPv6Ready" certified kernel. The software is "ready", but the OS also needs to be ready.
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How much RAM is considered normal use? I am seeing anywhere between 100M to 250M
seems like alot of difference.
"It depends". For a 10-user installation how have to deal with maybe 40 MB, for 100 users it can go into the 100-200 MB. The number of calls also does matter.
Fortunately, memory is cheap these days and usually not a problem.
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[9] 2009/02/15 11:00:46: Resolve 3243: url sip:tpip-p003s.trainpro-ipnox.net:5060;transport=tcp
[9] 2009/02/15 11:00:46: Resolve 3243: a tcp tpip-p003s.trainpro-ipnox.net 5060
[6] 2009/02/15 11:00:46: Could not determine destination address on 3243
Smells like a DNS problem to me... Maybe you can use the IP address in the outbound proxy instead to see if that is the problem. Keep the DNS name in the domain, so that Exchange does not complain.
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Sounds like you've been able to replicate the problem. I'll await a fix.
Can you try it out? We can make a test build for CentOS.
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Looking for an update on this issue.
It looks like we have a race condition here. When a message is being recorded it takes some time (ms) until the file made it to the file system. If the check if the file is in the file system is quicker than the file system write, the PBX might really think that there files is not there and it will delete the message.
Workaround: Hard to say. Probably we need to put in a extra safety belt that does not delete "young" messages.
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But it needs to. if my trunk is full i want it to go back to the dial plan,, not give the phone a 404 and give up.
I agree. Lets see how quick we can incorporate it.
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\*([0-9]*) replacement is *\1
Not sure, but you might need a pattern and replacement like this:
Pattern: (\*[0-9]*)@.*
Replacement: sip:\*\1@\r;user=phone
A lot of baskslashes! IT world meeting telephony world...
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do I need to put something in the dial plan for it to send the sip:xxxxxxxxxxx@localhost to the PSTN port?
Check if you can make an outbound call to the number from that phone. If that does not work, then click 2 dial will also not work...
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Please note that the users can access the PBX only by using the IP Address of the PBX server.
That will not work. Multiple domains require either an outbound proxy in the SIP phone or proper DNS entries.
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When you are listening to a message you are supposed to be able to press 1 and copy or forward the message to another mailbox. When I try doing this I just get dead air, or it goes back to the main mailbox menu. I have a client that wants this feature, but I need to get it working first. Can anyone help?
No, when you are listening 1 is rewind backwards. You have to press "6" to move or copy it. Then you get into a special menu. Make sure you have the latest WAV files; there are some new prompts; if you don't have them you'll hear digital silence (there will be a LOG message about this).
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I am not following, there is only 1 service flag field in the autoattendant, and one night service field.
If you seperate by spaces, the first service flag corresponds to the first night service number, second to the 2nd, etc.?
Yes. See http://wiki.pbxnsip.com/index.php/Auto_Att...t#Night_Service. A dirty, but flexible trick!
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I tried it here. For me it works - of course... However, I was using snom version 7.3.14 and there all SUBSCRIBE messages now show Expires: 0. Needed to change that to 3600, then it was much better.
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I have the same problem too.
Sometimes wy users are hearing fully random dtmf tones during the conversation without anyone pressing a button.
What phone type? What firmware?
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First thought is, 24 extensions 10 thru 34 with service flags 50 thru 74 1 for each hour of the day, calls are redirected to cells based on the assigned call flags. Sounds like a maintenance nightmare after the installation to manage cell phone entrys. Perhaps a simple ACCESS web page that can SOAP update the cell phone numbers... Just a first thought...
I would just use one auto attendant and 24 service flags. One falg for each hour. Then use the night mode feature to distribute the call to each cell phone:
Service Flag Account: 7100 7101 7102 7103 ...
Night Service Number: 9787462777 9787462778 9787462779 9787462780 ...
You can even make exceptions for the weekend and for the holidays!
trunk doesnt failover
in Trunk Setup
Posted
You need a 3.3 build for that. If you want to try - let us know what OS you prefer.