Vodia PBX
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Posts posted by Vodia PBX
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What does the underline part above mean? isnt the mixing done at the pbxnsip level? why would the phone need a good echo cancelaltion?
Sometimes it is also called "whisper mode". That is because when the teacher talkes too loud, the phone of the agent might have a little bit echo which would be audible by the customer. Maybe today the problem is not as big as in the good old analog times when there was no echo cancellation on a handset at all.
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Is there a global setting to limit the number of registrations per extension?
Now there is (version 3.3).
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I'm not linux so you may need to explain what top is.
Top shows you what processes are active, how much memory is used and all kinds of other interesting stuff about the healthyness of the system. The command line is just "top" <enter>.
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Anyone?
Well those templates are moving targets. But no secret. So feel free to take today's snapshot!
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I have the same problem that the cs410 will run for a couple minutes then I can't even ping it.
Seems like 10 minutes or less.
After a reboot it works for about 10 minutes and becomes unpingable again.
(initially i also noticed that i needed to plug the cat5 cable into the WAN port for the unit to get an ip via dhcp.)
its a black unit. Just got it. I upgraded it to the latest firmware/image.
Again very strange. Can you SSH in and run top?
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[7] 2009/02/06 15:28:34: Close TFTP port 4833
Did you select HTTP in the server menu of the phone? It should not generate a TFTP link at all...
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Status>Calls
I now get numbers under Action colomn. Waht does that means? (-29.5)
Start From To State Action
2009/02/04 08:19:58 Guylaine (73) 9xxxxxxxxxxxxx [xxxxxxxxxx] connected X (-29.5)
values are different between calls and lignes used. Is this audio Gain based on the Audio test from Logs?
Oh that is the call volume. This is for adjusting the gain. This does cost a lot of CPU horsepower, you better turn this off if you don't need it. It is in the Log section of the admin mode.
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for some reason i am getting the following error when trying to provision the phone remotely!!
0101015549|copy |3|00|tftpLib error: tftp transfer failed: error 0x4b0008
0101015549|copy |4|00|Download of '2345-11402-001.bootrom.ld' FAILED on attempt 1
For remote devices, you usually have to use HTTP because TFTP is not a very NAT-friendly protocol. See http://wiki.pbxnsip.com/index.php/Polycom#...ng_Provisioning on how to do that.
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please see attached wireshark trace....im trying to attach it!!!!doesnt allow me to!!!
Send an email to support@pbxnsip.com...
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I am on PBXnSIP version 3.1.2.3120
Upgrade to 3.2... It could be that you are dealing with a problem of version 3.1 - see http://wiki.pbxnsip.com/index.php/Release_Notes_3.2.
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I dont think it is a wireshark issue as the logfile does not show that it is playin Ringback.wav...i will show below the logfile.
5] 2009/02/05 09:18:47: Identify trunk (IP address and DID match) 2
[7] 2009/02/05 09:18:47: Set packet length to 20
[6] 2009/02/05 09:18:47: Sending RTP for e74464f9-f38f-11dd-bf57-d03f7f763324@pbx#ae480c5bb3 to 10.0.10.16:14010
[5] 2009/02/05 09:18:47: Trunk T1 sends call to 8401 in domain localhost
[7] 2009/02/05 09:18:47: Attendant: Calling extension 8401
[7] 2009/02/05 09:18:47: Set packet length to 20
[7] 2009/02/05 09:18:47: Call e05e8ec6@pbx#3380: Clear last request
I HAVE LOOKED AT OTHER PBX'S AND THEY SHOW THE FOLLOWING LINE WHICH THIS PBX IS NOT SHOWING
[8] 2009/02/05 09:20:32: Play audio_en/ringback.wav
why would my phone system stop playing this back? system files corrupted?
Well, the reason could be that the system is not supposed to signal that the other side is actually ringing. Or something simple like the log level is set to 7 and the PBX would not show a message on log level 8. I assume the file is in the file system. Corrupy file systems are pretty low probability these days.
It could be really something that the phone is really not ringing or that the PBX does not know about it. A Wireshark trace from the PBX server will show this.
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Can there be a setting that says after X seconds of ringing put the caller back in queue (priority call) and log the agent off? that way the customer gets the next agent in line.
I see the downside now the agent comes back and they dont know they are logged in..
It would be better IMO to have this setting as an option. I prefer customers to be happy and fire the agent
Automatic log off sounds interesting to me. There are already ways to send the call to another location, including the same ACD. That might be a temporary workaround. Especially if is practically only one agent in that queue, so logging that one out really puts us into trouble.
The problem that we are more and more dealing with are groups with very little staff. Very! So very that the customer chance to meet a real person is so low that we need to think about ways to pretend there is a real person. It is a challenge.
Unfortunately, if you fire the last agent that will not make the customer more happy...
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If this error continues to occur, please contact your network administrator. The network administrator can use a tool like winerror.exe from the Windows Resource Kit or lcserror.exe from the Office Communications Server Resource Kit in order to interpret any error codes listed above.
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I am not an OCS expert, but that sounds like you need to tell OCS how to find the destination.
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I am automatically provisioning the phones with an xml file. I have the TFTP password set to ALWAYS. When I loose the settings I am also unable to log into the web and it fails to register however, I can ping it and see it in wireshark requestin aI have noticed that after I provision the phone you log into the web interface with the phone username and password which is awesome and my directory works as well. I am using the Snom 370.
Hmm. Are you on version 3.2?
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Ok FINALLY! I was able to provision the phone and get the address book to load. But if I rebbot the phone everything fails. If I look in the wireshark capture I can see this:
Subscription-State: terminated;reason=noresource
I am not sure if this has anything to do with this or not. If I reset the phone to defaultsd it works fine. Any ideas?
No. That should not be the problem...
Do you automatically provision the phones? What is your password policy for provisioning (See Admin/Ports/TFTP). Maybe your phones don't get the password after the first provisioning.
What phones are you using?
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Hi guys,
First of all, this software is great! Glad that I was able to find this. Also kudo's for the community on this forum!
2nd, I'm currently Experiencing a outbound problem when dialing.
For example, here (Holland) we have 10 digit numbers.
I can manually dial my phone at 0123456789, and it will work. But, when I receive calls from outside, they are registered as f.e. +31123456789 (0031 country code), and I cannot return those calls, Office Communicator 2007 R2 immediatly gives me a
So the first 0 becomes a +31. When i callback that number, Office Communicator 2007 R2 "call not completed or has ended" error.
When I expand that error message i get this (Office Communicator 2007 Error ID: 403, Your user account is misconfigured on the Office Communications Server and you have attempted to make a call or join a conference. You are not allowed to do this operation on the server side.)
To be honest I dont think that has got much to do with it.
Do you guys have any idea on how to resolve this issue?
Try to set the country code in the domain to "31". On the trunk you can set how the trunk should represent numbers to the outside world, e.g. with plus or 00 at the beginning. Make sure you are running 3.2, I think that option was just recently added.
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I am ruynning version 3.1.2.3120. I just tried creating the trunks for the first time. I may try to edit the xml manually.
I would move to 3.2 anyway.
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We found out what happened. The problem was caused by one of our SIP providers that delivered instable services for a couple of days (with the trunk connecting sometimes, sometimes not). If the auto attendant transfers the call to a SIP line that's not available for a short moment it just hangs itself up. Now we selected a different SIP trunk for these calls and the problems is gone.
Hmm. So that means that there are call that just hang and don't get out of the system (maybe after two hours or so)?
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I just got a CS410, setup was nice and easy works the way I want it to. The problem I am having is this ... Call in to the CS410, call goes to a hunt group, phones ring, if nobody answers call goes to to voice mail on an ext. Simple setup. I can only get one call in, after that the system becomes very slow, can't open the web interface or ping the PBX. sometimes I can get two calls in before it becomes unresponsive. The only way to fix it is a reboot of the system. I tried to run a top via ssh but ssh dies so I can't see what process it is hanging on. firmware version is 3.2.0.3143 ...
If you cannot ping the system anymore we are talking about a problem at the lower levels. Maybe you can log in through SSH right after startup and then start top to see if we have a problem with the memory, CPU or something else looking strange. Anything else strange? If the network in good shape?
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That is the problem that I have. I don't know why the PnP configuration does not work. I have followed the Wiki. When I boot the phone and log into it I can get pbx to assign the mac address to the users extension but the dir (phonebook) does not work. I have 2 deployments next week and another in a month or so. So I need to get this ironed out.
The firmware issue is solved. After downloading the original firmware the phone is working again! Now if I can get that phone book issue taken care of.
You can still perform a TFTP software update of the phone through the bootloader. That should also reset all configuration, including the one that is causing the problem.
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Hmm what does this mean from the log file. I can see it deleting some of my co lines.
[0] 2009/02/05 08:17:44: Database sanity check: Deleting coline 15
[0] 2009/02/05 08:17:44: Database sanity check: Deleting coline 16
[0] 2009/02/05 08:17:44: Database sanity check: Deleting coline 17
That is not good. It means that those CO-lines are not properly associated with a trunk I guess. Could be from older versions, or they could be lingering from an unsuccessful attempt to create them.
Anyway, this log has nothing to do with the number of CO-lines that you can have on a trunk.
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On a client's system, I have noticed that calls that are forked and picked up at a user's mobile phone are listed in the system's call log, but do not show up in the user's web portal call log.
Call trace emails do not seem to get sent either (but I haven't tested this as thoroughly.)
Is there a way around this problem?
My customer is running 2.1.14.2498, and cannot upgrade to 3.X at this time due to very heavy use.
Is there any chance to set up another system for testing (3-minute key) and see if the version 3.2 solves the problem? Then in a seconds step we can see what we make out of this.
Unfortunately, the CDR logic inversion 3 is quiet different from the logic in version 2, so this will not be an easy patch.
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I have a an Installation that moved from a private IP to a public IP to allow external user to register phones from their house.
My problem is that all phone register but the one outside of the company location do not play any audio at all.
I've tried to check if ports are open/close and I can see that 5060-5061 are open but 69-close 49152 to 64512 (that are used for audio if I am not mistaking) are closed event on public IP?? From the Internet service provider ? is that possible?
Does that make sense and can it be the reason I get no audio at all?
It could be a problem with the routing table. It is not uncommon that the OS chooses a different route when you add a second IP address. Sometimes the NIC with the private IP address has a higher bandwidth and the OS thinks it is wise to send all traffic through the NIC with the higher bandwidth. What is the output of the "route" command from the command line?
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How many co lines can I have a single trunk? I have a mediant 1000 that has a T1 with 23 channels. Do I need to create more than 1 trunk group. I see in my single trun group that I am missing a handful of the lines I know I put in there. Thank you
There is no limitation built-in. 23 Should be no big deal.
agent ringing timeout
in Agent Group Setup
Posted
The agent group has a couple of settings that redirect the call out of the queue ("when ringing for ..." and so on).