Vodia PBX
-
Posts
11,069 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
HOW CAN I INTEGRATE OUTLOOK2007 WITH CS410 DEVICE
TAPI: http://wiki.pbxnsip.com/index.php/TAPI_Service_Provider
Apart from that check out the email support.
-
What is the format for setting the start times available and not available?
It is the same as used in the Service Flag, see http://wiki.pbxnsip.com/index.php/Service_Flag.
-
I have even checked with my ITSP provider.... they are on Asterix..... and they are using the RFC 2833 for DTMF.... but tis doesn't work....
We had a case where Asterisk was sending RFC 2833 DTMF, but extremly short ("Blitz DTMF"). It was so short (0 ms) that the PBX never had the chance to transcode it into inband DTMF. Maybe that is the problem here as well.
-
I've had a play with VLC, but all I get is screeching when I put a call on hold, I suspect I've not got the codec/mux settings correct, but I can't see what they should be set to. Does anyone have definite working settings?
It being used on the CS410, the music in jack streams data on the MoH port. It uses (mono) G.711.
Maybe do you have a Wireshark trace of the VLC stream?
-
We have 20+ domains on 1 FS - our plan was to move the domains one or two at a time - using the "backup domain" utility - which turns into a domain.tar file.
We manually moved MOH files over - because they were not domain specific, and understood.
But the greetings for extensions - did not come with them... The AA recordings, and some name recordings came with - but not the personal greetings.
Does that help?
Ah. Yes that helps. The personal greetings should be moved between the same versions. When we introduced the possibility to have several personal greetings, the file format changed. That should happen automatically when performing an update on the same server, but it is not supported when moving a domain from a "old" format to the "new" format.
-
I have a PBXnSIP installed on server (windows server 2003) and the problem I have is that the service goes down and I have to manually upload.
Start the services manager (Run services.msc) and edit the properties of the pbxnsip service. There is a tab for service failure, you may choose to restart the service after the first failure.
-
We have a customer who is having trouble with their conference room. They have an inbound VoIP number routed to them which is aliased to a conference room. When there's more than a couple of participants, audio drops out every 3/4 of a second or so, it's really regular. The PC it's on is only running at 2-3% CPU, and 4-5 8K streams shouldn't saturate a 2MB SDSL pipe.. Any ideas or possible things to look at?
We had a router that was running out of CPU when the number of packets per second was getting too high. Wireshark will show this.
The other thing is processor affinity, virtual machines and swapping. Maybe turn off logging to make sure it is not something stupid like the file system.
-
Please let me know if I need to open a trouble ticket on this for faster response - This is critical to our upgrade path from Windows to Linux - and we have 7 Feature Servers with multiple domains to backup and move.
Thank you.
Okay, let me restate the question... Are you moving the whole server or just a domain? You are really making a ZIP of the whole directory and unzip it on the Linux server?
I verified case and slashes - they should be fine. Can you tell me what is in the extension/nnn.xml (look for *.wav) and if that file exists in the recordings directory?
-
I have my network setup with all voice on VLAN 3 The problem is that the autoprovisioning feature keeps removing the VLAN settings from the phone, and it looses connectivity. I was told that I can go into the generated folder, and manually edit the file for the phone, however PBXnSIP overwrites my setting changes to the file. I am forced to manually configure the phone. Is there a fix?
There is a VLAN setting in the PnP parameters of the PBX, did you try that? I saw in the release notes of the snom 7.1.33 that the new version will take one more reboot cycle in order to get the VLAN parameter correctly, maybe that was also part of the problem.
-
A person leaves the company and I want to reuse the extension number. I need to have ALL traces of that users messages and their voice mailbox message deleted and be able to assign that extension to a new user.
Do I have to delete the extension OR is there a way to reset that extension so when i assign it to a new user, they can go in and hear the message to set up their mailbox and have no traces of the previous mailbox user.
The easiest is to delete the extension and then create a new one. There is no explicit reset button.
For hospitality environments, we added a SOAP request so that the PMS can trigger this action when the guest checks out.
-
Does the video portion of it use our bandwidth? Or is the video portion point-to-point / phone-to-phone direct?
All media is subject to the SBC functionality of the PBX, which is very useful if you want to send FAX from one device behind NAT to another device behind NAT. Maybe a video client can also be behind NAT. The good news for video is that the packets are large and the frequency is low, and the CPU has not so much to do shuffling the packets back and forth between the OS and the application.
-
I just zipped a domain from one FS running 2.1.6.2450 (Win32) and moved it to a new FS running 2.1.8.2463 (Linux) - Most things worked correctly - however the voicemail Greetings were gone, and must be re-recorded. We experienced this on a domain zip - move from Win32 to Win32 as well. Could you please identify the problem and implement a patch fix? We need this feature, as we are rolling to Linux - we have to zip and move.
Hmm. The only thing that comes to my mind are case-sensitive file names. Anything in this direction? The other things are backslashes - anything there in the XML files?
Maybe we need a short script that patches the XML files accordingly.
-
If phones are working fine with Asterisk, Communigate, Brekeke, SER and other application and only one which is not working is PBXnsip,- the problem is in this application. To be honest we did not test other GS device then we have.
I don't know... I don't think it is okay to say the device is crashing and they don't have any plan to take care about it. Other vendors take crashes pretty serious.
I think the only thing that we can do is trying out another version that is not crashing the phone.
How about second issue? Why 2.0.3.175 is working and 2.1.8.. is not (one way audio)?What you can try is disabling the pass-through RTP. There is a global setting called "allow_pass_through", if you set it to false then it will not simply pass the RTP through, but run it through the jitter buffer. See http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set this setting.
-
No, I am not using "localhost" domain. Also I've tried to use "Outbound proxy", STUN, different set of codecs etc. and to not use...
Why Grandstreams and Netgears only?
Well, a device should not crash if there is something unexpected. Did you contact Grandstream? I think if the device is crashing it should be a high priority trouble ticket and you will get a new firmware soon.
We are using an ATA in one office (Grandstream HT-502 V1.1B 1.0.0.44), and it was so far pretty stable and also T.38 is working fine after we disabled UPDATE support on the PBX.
BTW don't use STUN, it is instable and IMHO not suitable for enterprise communication.
-
This appears to be a reoccuring theme in the latest releases - Do you believe this will be rectified prior to future releases, or will be something that we will have to do for each upgrade to ensure the references are stable?
The problem is the underlying data model. If create a service flag, it will get a number. Then if you reference it from the extension web page, everything is fone. When you delete the service flag and create a new one, that one has a different number, even if the name is the same. Then the reference gets broken.
That is perfectly normal to a computer programmer, but not for a end user. We think using more Ajax can address this and make end user's life easier, but we want to stay away from hardcoding the internal database relations and resolve conflicts.
-
Are you using outbound proxy? Some devices have problems is the domain name is "localhost", that may be the problem with the Grandstream devices as well.
-
I am posting to see when video might be available for Pbxnsip. Anyone with any info, please let me know.
The status is that video works if you first have a audio-only connection and then switch to video (with a Re-INVITE). It is all because of T.38, which is also kind of video (pretty static, though).
-
i've got 7 soundpoint ip301 phones that need to be setup with the cs410wp. i have a piece of paper with "step-by-step" instructions on how to do it but imo it sucks or i'm stupid or both. does anyone have any pointers they'd be willing to give out? any help is much appreciated. thanks !!
For the PBX:
http://wiki.pbxnsip.com/index.php/Installa...and_Quick_Start explains some general information.
http://www.pbxnsip.com/templates contains a 10 extension configuration that you can import into the PBX to get jump-started.
For Polycom:
http://wiki.pbxnsip.com/index.php/Polycom has some infos on how to set these devices up. http://forum.pbxnsip.com/index.php?showtop...tart=#entry4014 is a very interesting post on how to set the Polycom phones up.
-
I have problems registering to trunks after upgrading to 2.1.9.....
This problem goes away when I downgrade back to 2.1.7.....
That does not ring a bell... Any specific information, SIP trace or so?
Redirection to cell for with service flag set or clear have no effect.Calls are still redirected to cell regardless of the state of the service flags in extensions..... hunt groups works fine...
Try hitting the save button again. Maybe the reference got broken somehow, saving it would restore it.
-
I want to know if anyone has experience with the TELES appliance and the software is a cut down or regular version of the software. I want to make an immediate decision.
It is a regular version, no feature excluded. The limitatons are the memory, for example call recording will be difficult. But the good news is that it comes with a great PSTN gateway.
-
So if there are leftovers from previous versions, how can I be sure, as administrator of the server, to have a "good" upgrade? Usually I will install over a previous version so as not to lose any of our users settings, etc.
We try to make only bugfixes in the 2.1 branch. New features (even if some people don't like it) go to the head branch (we will probably call it 3.0 then).
-
I am running 2.1.8
The Cisco is POS3-8-9-00
Here is the packet
Probably the Cisco did not like the http in the alert info. Can you give the attached ringback.xml a try? You need to restart the service for that or load it through the special web interface page (Reload Configuration Files in admin/configuration).
-
No one has answered my other post about not being able to autoprovision the phone's buttons since doing so knocks it off of my voice vlan, so I am forced to manually provision the phone.
Oh, but there is a setting for the VLAN in the PnP settings (admin/settings). Maybe you just need to specify the VLAN there.
-
The problem is they might not come back for hours, or might not make a call for hours/days, and by the time the callback happens, the person forgets that they ever requested the camp in the first place.
Okay, that makes sense, but then maybe the solution would be to have some kind of timeout with the callback (e.g. one hour).
Transfer from Cell Phone
in General Setup
Posted
Nope, that is not possible... The call is on a trunk.