Vodia PBX
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i am having a problem relay voice mail to my exchange 2007 server.
both are on an internal network of 172.16.1.0/24
what are the correct settings i should have on my exchange server to relay email properly?
Well, you have to use SMTP. I know it works with the currently published version, there are several installations talking to Exchange with SMTP. But I am not the big Exchange expert, on the PBX side you just need to point the SMTP server to the Exchange server (that part is easy
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I am seeing the same thing with people going over 3 stages when using the mailbox escape account
Hmmm. There is a setting "max_loop" (default should be 10). Maybe that is the problem here?
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I would suggest that in the individual extension configuration web pages that a next account, and previous account button be added. This feature would dramatically speed up mass changes. For example right now if you want to change a feature for a bunch of accounts, there is a lot of lost time clicking and navigating. If you wanted to change the dialog permissions on a bunch of accounts, right now you need to save, and hit list, find the next account, click on the edit icon, and click on the registration tab, and then enter your value. If there was a previous/next button, you could hit the save button, then hit the next button, and you would be to the exact same location on the next sequential account number, and ready to enter your value.
Well, what we could do is having a link in the files dom_ext*.htm for prev and next.
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I just tried it on a different server same thing.
Try this scenario. Auto attendant then press 1 for a hunt group. ring a few phones for a couple seconds then time out at the auto attendant again. then press 1.
after 3 events (or 4) the system will hang up. no matter what events (looping or not looping)
Oh, do you have a hangup timeout on the auto attendant? Maybe it is still active from the first AA and then kicks in when the PBX is back in AA state?
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In the registration tab, it would be nice if i could put "MAC MAC" so I could do PnP with 2 phones registered to the same extension.
yea. But we need to split up the table, in order to make it possible to index it. This will be a little bit messy, that'S why we don't have it yet. But it is noted. -
bump. will this be fixed in the next release? people are complaining about it.
The text "Service Flag" is misleading (does not say anything). We'll rename it to "Specify time when system calls the cell phone", hopefully that makes it easier to understand what the PBX does.
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UPDATE: yes it works after restart
You can also load it through the webinterface (admin/configuration at the bottom). Then you don't have to restart.
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When an external call is connected and you want to transfer this to another internal extension with announcement, you place the external call on hold and push the internal BLF button for the internal extension. Then you can announce the external call and when it is ok to transfer you push the transfer button on the snom phone and press "V" to transfer. Then the BLF lamp from the internal extension remains blinking like it is on hold. Also the PAC is showing that the extension is on hold, but the extension is talking then to the external caller !
Okay, that sounds reproducable. We'll track this topic and post when there is a fix available.
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Thanks, I have just installed this version.
I have found the setting in the webinterface. It seems exactly what I need but it looks like it is not working.
I have configured it to "1" so at least one agent should remain logged in, but they are all still able to logout.
And there is a problem with the codec configuration field, it is changed to a table with up and down buttons, but the configured codecs are not listed anymore, the table is empty. I have tested this with IE7 and with Firefox with the same results. I see the following error in the logfile, perhaps that is the problem:
Web Server: File codec_selection.js not found
Ops, will be fixed in the next build.
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I am trying to setup an Aastra 57i in a typical squared line button configuration. For those of you who are not familiar with this old PBX term, it would be so that the line buttons on the phone corrispond with CO lines on the system. This would enable someone to take a call, put it on hold, and by remembering what line it is on, go to any other Aastra phone, and press the line button to pick it back up. I have done this successfully with Snom, is the Aastra 57i capable? If so how?
AFAIK Aastra does not support "buttons", only "dialog" state (sometimes calls "BLF"). With that CO-line emulation is not possible. The only thing that you can do is use star codes, e.g. to park and pickup calls.
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I have an incoming call that rings an auto attendant and they press 1 which rings a hunt group and if nobody answers the final stage is to go back to an auto attendant and it fails.
If i call the hunt group and bypass the first auto attendant it works just fine.
UPDATE: seems like you can only do 3 system 'thing' before it kills in any scenerio.
Works for me... I assume you are using a direct destintion for the "1"?
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How many conference users can a 3GHZ PC handle? no other traffic just conferencing server with many rooms.
That's one of those questions which are really hard to answer. There is a FAQ on this topic (see http://wiki.pbxnsip.com/index.php/Hardware_Requirements and also http://wiki.pbxnsip.com/index.php/Performance_Optimization). In the end, you need to try it out on your hardware.
Also, it seems that is heavily depends on the operating system and all the firewall stuff going on. Looking at every packet does cost performance as well!
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I told my colleague Giorgio ( Gunny ) had already opened a topic for this problem.
He is still waiting for answers because it seems that does not work.
This feature is very important to choose the product and sell it to our customers. :-)
this is the topic link:
We did take a first look into it, but could not identify the problem. But it is not forgotten...
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thanks, but dongle USB in italy it affordable ?
Well, the dongle needs to be shipped from the pbxnsip sales office (UK), that should not be a show stopper. The dongle has the benefit that you can plug it into any computer, not matter what MAC. That is a big asdvantage in case your server hardware crashes and you want to continue operation on another machine...
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Wow, that would be very nice, I am using the Win32 version.
Is there also something changed for the strange BLF behavior as I described in http://forum.pbxnsip.com/index.php?showtopic=1492 ?
The latest and greatest currently is http://pbxnsip.com/protect/pbxctrl-3.1.0.3039.exe; not released yet but you'll see the progress!
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Are there others who have seen the above problem?
Is there a solution for it?
First, I would not use version 7.3.7. 7.1.35 is the best version for the pbxnsip right now.
Second, are you really using dialog state or are you using buttons? dialog state has many many problems and on snom phones buttons makes your life a lot easier.
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can anyone explain to me why when i switched the default gateway of the CS410 to the lan port instead of the wan port it stopped giving me access remotely over the WAN port?(both for setup AND remote phones)
i have a numbe rof setups like this when the default GW is the LAN port (for redundancy with a dual wan router) and i have no issues getting in remotely using the static wan ip address.
using version .3023 of cs410 firmware.
This is a general Linux routing problem. It is also kindof black magic to us... We did nothing special in the CS410 compared to other Debian distributions. You can check the /etc/networking/interfaces file to see what we present to the OS.
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I have a user who somehow created about 25 out-going messages he could not delete them, so I went to the recordings folder and deleted them for him. However, when he goes in to choose which message to use it still thinks they are all there, it says for.. (then nothing) press 2 ect ect.. How does he delete them all and start over, he has a ton of saves messages so I would rather not delete his ext and set him up again, there are at least 4 others who have the same issue.
The key problem is that the recorded message is treated as call recoding message. That also had the problem that the system would send an email when the recording is finished. We have to split this up into message recording and call recording.
The other thing is that we check if the file still exists before counting a message. That would have the nice side effect that deleting WAV files from the file system will automatically (eventually) delete also the associated message. That would be a very easy way e.g. to remove messages after 14 days.
Will be included in the next release, 3.1. Preview available on demand...
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hi there, i was curious to know how someone external (with a polycom phone and headset for ocs) would access ocs and pbxnsip that is internal. in this guide at the bottom http://wiki.pbxnsip.com/index.php/Office_C...ications_Server it has a diagram showing a sample setup, the only thing external is the ocs edge server. we are currently testing the same kind of setup and would like to know how we would go about setting up remote users to access (pbxnsip or OCS?) to the DMZ (What server should be here) first?
Well, if you have Polycom phone then you would just register the phone with the PBX. You can use the guidelines in http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses for registering the phone.
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During phone boot, the phone can't find it's boot server & config files. I'm getting an error message for:
1. "Can not contact boot server. Using existing config."
2. "Error loading xxxxxxxxx.cfg"
After posting error messages, the phone re-boots.
My questions:
1. How do I establish a "boot server" to address this phone?
2. what are the parameters and instructions for setting up the list of "config" files?
Did you check http://wiki.pbxnsip.com/index.php/Polycom? There you see what you must do on the phone in order to get it working.
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Is it possible to create that feature "Have at least xxx agents in the queue"?
We already had some filled queue's without any logged in agents....looks not very professional to our customers!
Yes, that will be in 3.1. If you want a preview, let us know what OS you have.
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How we can buy a license?
- Ip address
- Mac address
- Dongle USB
it's correct ?
Yes. But: We do IP addresses only for hosted servers. A MAC address must be a real MAC address (no virtual machines).
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In the "EDIT DOMAIN" section you can limit the calls to that domain to say 3. Cant you add a link below that that says overflow?
So when the call limit counter hits 4 i could specify 8100 (voicemail for 8100).
Hmm. That means whenever there are too many calls in the domain you send them to the mailbox?
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We have tested the pbxnsip with our voip platform.
We have configured the AA in order to recognizes the FAX and redirect it to account.
Now, the account is a sip Telephone (for test).... but not rings.
nothing ... not works.
Did you turn "inband detection" on (system settings)? See http://wiki.pbxnsip.com/index.php/Overall_...gs#SIP_Settings. You should see something like "[6] Received DTMF F" in the log file.
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yea. But we need to split up the table, in order to make it possible to index it. This will be a little bit messy, that'S why we don't have it yet. But it is noted.
Recording Extension does not work
in Call Recording
Posted
Are you recording all calls or just the calls to that extension? Because after a redirection, the call technically does not go to the extension any more.