Vodia PBX
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Posts posted by Vodia PBX
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Off topic to pbxnsip. but wondering if I can push out an XML page to the snom 360/370 from a application server (non-pbxnsip)
You can do that. Just specify a ActionURL on the phone that points to the application server.
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[5] 2008/10/10 05:21:31: Attendant: Redirect to
That looks unhealty. After the "to" there should be a number, not the "empty" string. Maybe the redirection number is something funny like a space character?
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i h ave a problem please help me, how i can transfer a call that cames from sip trunk the trunk change the real numeber to an extention of the gateway and the call dont terminate because the pbx change the real numnber to a extencion are there any patterns for trnuks o what i have to configure i dont need to change the real number the did to cid the user will call a number and that munber must arive to other gateway , well in the meadle is the pbx and that is changing the real number. please help me.
Ehhhh... I have difficulties understanding what the problem is. Can you make an example?
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This seems to be the same issue as indicated in:
http://forum.pbxnsip.com/index.php?showtopic=1254
PBXNSIP, any ETA?
A beta is already available. What OS do you prefer?
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TLS
in VoIP Phones
OK, I've found thishttp://wiki.pbxnsip.com/index.php/Getting_...lid_Certificate
but I don't think this would help, right?
Well, counterpath is strict with the certificates. As a rule of thumb, your Web browser must be able to go to the web server of the PBX (using https), without complaining. You can do that by importing e.g. the cacert.org root certificate into the Internet Explorer. I did that some time ago and then the counterpath softphone worked fine.
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After upgrading to 3.0 the outgoing numbers are wrong. When i call from extension 45 it shows +45 as caller when it should state the real number 0842013045. How do I correct this?
Instead of tel:alias you should use the ANI fields in the extensions in version 3.
The + sign may come because you selected a PnP scheme for ROW. If you don't need it, you may turn this off.
Or just turn the flag off "Interpret SIP URI always as telephone number".
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Huh, help?
The L&G is http://pbxnsip.com/cs410/update-3.1.0.3032.tgz. maybe give that one a shot. It contains also a first version of the hangup detection performed by the PBX as a trunk setting.
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So when I press DND on my SNOM phone, it sends pbxnsip the DND *code.
Would it be possible to light the LED on all the phones so they have indication that the phone is in DND? be cool if it would wink but dont know if that is possible.
It does that - if they are registered to the same extension. If you want to monitor other extensions, you (currently) need to use the monitor extension (extended) mode. However the phones don't display that yet.
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ok i am confused. The allow flag is ok. So i enter my extension then pin and i should press **86?
No just *86. The other star in the beginning is not neccessary because the input is empty in the beginning and does not have to be cleared. Maybe just give it a shot.
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I enabled the Global radiobutton and the Accept Redirect on the trunks.
Each domain has it's own trunk because echt domain is another company's so they have their own outbound number.
And what is two domains both contains extesion number e.g. "100"?
Or do I need to set some other thing in the trunk config?
Edit: I also set the "tel:xxx" alias names but... nothing
The topic of sending one call from one domain to another is a compliated one. You really need to use a trunk that goes out from the PBX and another trunk that comes into the PBX. I would setup two trunks, both of them with the outbound proxy sip:127.0.0.1 (or sip:[::1] for IPv6), one of then for "inbound" and the other one for "outbound". The inbound should be a global trunk, so that you can send the call to the right domain from there.
The tricky part is to find out if the destination is local or not. You can do that in the dialplan, if you have just a few numbers. For example, if you have a central secretary service you can bypass PSTN and send the call directly back to the PBX.
If you have a lot of numbers that you want to route directly between the PBX, you probably have to employ DNS to resolve the final destination. If you run your own DNS server and you are able to set up ENUM entries (which is easier than it sounds), you can first use a trunk that performs a ENUM lookup and perform a failover to the next trunk in the dialplan if the entry does not exist. If you get that working then you are in the seventh heaven of hosted PBX and you can scale that endlessly.
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ok but the box never rebooted and the customer called me and said their night service was active. I looked at the web interface and the date/time was wrong. so I sshed to the box and it was reset to Jan 1 at 3:00 AM. (which i assume means 3 hours ago the date reset.
the pbx was not reset. so my question is why did the box date reset? doesn't make sense.
I have 1 chronic box that doesn't this and 2 others so far that might have just been a fluke.
Oh. Setting the time on Linux has only limited effect. The PBX actually does it's own NTP and calculates the drift from the OS.
Changing the time on the OS screws up the callbacks. When you have a phone call and set the time on the OS, there will be a major RTP hickup. That's why we accept that the clock drifts and calculate the "real" time with this trick.
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Is it possible in the logo.gif field to put a rotating banner ad? like google adsense or something..
(execute code)
Interesting idea... AFAIK that should be no problem, because the animation is the job of the GIF and the PBX does not have to be aware about it.
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I am not linux guy, I am using WinSCP to look at it...I removed the log file, as it was getting pretting big and removed a lot wave files from the messages,
I guess the tricky part is this:
[5] 20080101000705: Trunk PSTN1 s
Seems then something really fatal happened before the library was able to flush the remaining characters to the flash. Are you doing anything strange?
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RedHat EL5
Try http://pbxnsip.com/protect/pbxctrl-rhes4-3.1.0.3031. Is has a global setting called "send_recording", if you set it to "true" it will treat recordings like voicemail. Maybe a workaround until we found a better solution.
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PLEASE.... Add the flag! In the meantime, is there a way for us lay folk to activate the feature again?
What OS do you run? We can provide a test image.
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It think the problem is not sending mials, PBXnSip sends ok the email with the attachment when is one voice mail, but when it needs to send one mail with the attachment with the recording just it does not.
That was supposed to be a feature... The recordings are now available from the web interface and are shown as such.
The motivation behind this was that you might want to receive voicemails as emails, but not recordings. Maybe we need another flag that tells the user what to do with the recordings, just like with new voice mails.
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im running 7.1.35
I have the system set to email me and save the email as read. however the light is always flashing and the voicemail on the phone says i have 1 new message. if i call the voicemail it says i have 20 saved messages.. then the voicemail indication is gone.
why.. is it not sending the proper messages?
It seems that the phone turns the MWI on even if there are "new" saved messages...
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It would be cool if when the incoming call is forwarded to your cell phone you could pres ## (or something) and be able to get system dialtone to transfer the call to someone.
Agree. Will be part of 3.1 and is currently under test!
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It would be nice if we can specify the voice processes to run on its own processor.. so the web and non-critical functions can run on a different processor.
Agree. We have plans to take advantage of the upcoming multiple-core processors, so that we can e.g. run 14 RTP processes on a 16-core CPU. That would make it possible to perform around 1500 transcoding sessions using G.729A. That should make the discussion about media-relay superfluous.
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Is there a way to setup a pre-AA greeting/message w/ a service flag before the normal AA greeting is played?
Client wants to close the office for bad weather. He calls in (from an external extension, cell phone or regular phone), records the new message "Hi today is monday December 21st and we are closed for snow". Set a service flag from the phone to play the newly recorded greeting before the regular greeting.
Right now there is a dummy extension setup - for weather press 4 and he can record that with the *98 codes. This works ok but not the ultimate goal.
Did you check the 2nd tab in the auto attendant? Check out http://wiki.pbxnsip.com/index.php/Auto_Attendant, it has just been updated!
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Calling card application wont allow me to dial a *98xxx why?
Well, one simple problem is that the '*' is usually used to clear the input. Only when you press the '*' at the beginning of the number then the PBX would take it.
The other point is that you need to check the flag "Allow calling accounts on the PBX". See http://wiki.pbxnsip.com/index.php/Calling_Card.
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Next step seems to be to load a test box with pbxnsip so that we can do a wireshark trace on an isolated call and see what is being sent between pbxnsip and the quintum pstn equipment.
Just don't use the 3-minute demo key
...Wireshark is pretty good in filtering out conversations. Maybe you can get just a huge Wireshark file on the ITSP trunk and then filter for the SIP and RTP traffic. Maybe be more efficient that setting up a second server.
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I see that 3.0.1.3023 has been released on the website. Can you please tell me if the recording/star code/email issue has been resolved in this release?
No, that did not make it into this release. In the meantime, can you double check that the license allows recording?
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It seems the problem is not between the phone (192.168.11.50) and the PBX. It seems that the problem is on the other side of the call. The phone sends media all the time, but the PBX is only sending keep-alive packets. That looks like the PSTN termination has a problem with media.
BTW try to put the version 2.2.2 of the Polycom into the tftp directory, maybe there is a problem with the firmware.
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Multi Domain trunks
in Trunk Setup
Posted
Well, there is currently no single "right" way to set this up. I simply depends...
As a rule of thumb: If you register trunks, you have to do that in every domain. If you can use a gateway mode, then you should have one trunk in a dummy domain. Then you can share that trunk amongst the different domains.