Vodia PBX
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we have a customer, where 5 users in an agent group receive calls.
Sometimes a user can receive e.g. 4 calls, and suddenly the phone goes on DND.
But the DND is not showed on the SNOM phone.
Only in the web-interface we can see that it is on DND.
First I did not belive the customer, so I took two phones back yo the office, and started to test .
I made 3 calls to the agent group. Worked. The phones rang and was not on DND.
Then hour later I tried again, and suddenly the one phone was on DND - but again, I could only see the DND on the web-interface. Not on the phone.
What can be wrong??
So the phone believes it is on DND? What version? Make sure that you are running 7.1.33.
There are two ways of dealing with this. The first is to reprogram the DND button as speed dial, use the same code for DND on and DND off and put that code behind the speed dial.
The second is to use the buttons and assign the DND "button" as DND. Then it should work similar to the speed dial, but also show the DND symbol on the screen.
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I tried the new firmware version on the PBX and I am still having the problem. Like clockwork, after the call is connected and on mute for 3 minutes it disconnects. I will try to get a capture of the traffic tomorrow to see if there is anything that will help there.
Are you using the 3 minute demo key???
Had the same problem with testing here
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i am afraid i need a link for the debian version
Try http://www.pbxnsip.com/protect/pbxctrl-debian4.0-3.0.0.2993.
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The first is is it possible to create a local database of caller ID names, so when the PBX sees a number, it inserts the name out of the local database, or can overwrite the caller ID name that comes from the phone company if the name exists locally. The reason I ask is we have some customers that receive a lot of calls from DID numbers at other companies where it will just show the caller ID name of the entire company, and not specific person who has the DID.
Sounds like you could use the domain address book for this?
Secondly in the case where the local provider only provides caller id number, it there a public database you can query to get the name? I would think something similar to reverse DNS would exist. If such a thing does exist, how could I get the PBX to query it?Such service exists, but you have to pay for it...
THe other thing is: For the PBX that is tricky, because there might be a significant response time and then the user will have the impression the PBX hangs. In SIP it is generally difficult to change the To/From-headers once the INVITE has been sent (or let's say the support from the phones is pretty poor).
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I wish it was taking my extension number, and sending it. It is sending my cell phone number.
Okay, just another idea. When you call from your cell to an auto attendant, and that number is in your extension, then the PBX gives you the option to call out. In that case IMHO it would use the extension caller-ID. Maybe that solves the problem?
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Can you give me an example for that?
the problem is, that in the dial plan I don't have the chance to set a hunt group as goal. I only can set extensions for direkt call, but pbxnsip don't accept a hunt group number as goal for direct extension call. I don't see a way, to tell him, if you get a call from OCS to 555 forward it to hunt group 555, it always goes to trunk xyz 555.
Okay, got it. That will be indeed difficult/impossible.
Hmm. The alternative could be to define a pattern in the trunk that tries to send the call to a hunt group, and if that fails it uses the dialplan. I am thinking about a pattern like this:
!555!555! ![0-9]*!\1!
Not sure if it really works, just a wild offline idea.
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We are getting ready for the next major release - 3.0. The image seems to be pretty stable now, the next action item missing is the release notes and then updating the Wiki. For those who can't wait and want to get the latest and greatest you are welcome to grab the Windows build at:
http://pbxnsip.com/protect/pbxctrl-3.0.0.2994.exe
The license keys for version 2 also work on version 3. When doing an upgrade, it is a good opportunity to make a backup of the 2.0 directory, just in case that you decide to roll back. The upgrade itself should work automatically.
Important things to consider when doing the upgrade (so far):
- The tel:alias semantics has changed. Tel:alias is only used for inbound calls, and only if the trunk is marked as a "global" trunk (that was a security fix that we also applied to version 2.14). For outbound caller-ID representation, we added a field ANI that can be assigned on account basis.
- We will post other important upgrade issues here as we find them.
- The tel:alias semantics has changed. Tel:alias is only used for inbound calls, and only if the trunk is marked as a "global" trunk (that was a security fix that we also applied to version 2.14). For outbound caller-ID representation, we added a field ANI that can be assigned on account basis.
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Is there a way to set the agent group so that if there are agents available, and their recovery timer has expired that the caller does not hear the initial announcement for the agent group, and they go direclty into ringing one of the available agents?
Well, then don't record the annoucement number 0, just record number 1 (e.g. *98123*0 and *98123*1). The special about number 0 is that is always played back, especially in the case you are mentioning.
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that sounds good, I exactly searched for that way.
But how do you call internal hunt groups from OCS user. In my configuration it allways looks for an outgoing trunk, even if I call a 4digit internal number.
I then implemented every 4digit in the dial plan as direkt extension call, but that does not work for hunt groups
Maybe you should reserve special prefix for hunt groups, extensions etc and then split it up in the dial plan.
The dangerous thing here are loops, make sure that such groups do not end up in endless call loops between OCS and the PBX. Those are really hard to deal with and instanteneously create the maximum number of allowed calls. Call it DoS if you want...
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It's actaully on a CS410 with the newest firmware release. If you have a patch you want me to try I can give it a shot.
Check out http://pbxnsip.com/cs410/update-2993.tgz and see if that fixes the problem.
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That would be great. I really need to get this fixed since I do a lot of calls this way. The way I have been testing this after the call I had yesterday is that i'll dial a land line and answer the call. I then mute the call on the pbx and within a few minutes the call will disconnect.
If you can give me any suggestions as soon as possible I would really appreciate it.
Okay, the phone really does not send RTP, even if "RTP Keepalive" is set to "on". At least in version 7.1.33.
But it does send RTCP. I think we can also use that as a indication that the call is alive. It might have problems with NAT, but I think it is reasonable to say that RTCP also is a sign that the UA is still connected.
What OS are you on? May we give you a image to try it out?
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I checked the version on the phones and it looks like they were running 7.1.3. I updated them to 7.1.33 and the problem still kept happening. I then updated one of them to the newest version which is 7.3.7 and I am still dropping the calls. I couldn't find anything in any of their release notes about this problem. You said that I could edit the pbx.xml to change a setting for the one way audio setting. It doesn't sound like the ideal option but if I wanted to test making this change, how do I edit this file?
If you can think of any other options that would be appreciated too. I don't think it is a Snom firmware problem since I have now tried it with three different versions.
Okay, I think we need to try that in our test lab as well, should not be too hard to reproduce.
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it the OCS Edition. I already found the problem: It only happens, if I use LDAP lookup for incoming calls. I don't know, if it happens generaly with all hunt groups, or if it depends on the number of mine: 0000. Is 0000 a problem for LDAP lookup in any way?
Well, IMHO the phone should never go belly up, even if complete nonsense has been entered.
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As we approaching version 3 release, we have spent some time in getting an image for the MAC world. Especially the MAC mini seems to be very interesting for running a SIP-PBX. For those who are interested, we have prepared a build at:
http://www.pbxnsip.com/download/pbxctrl-darwin9.0-3.0.0.2992
Feedback welcome. Maybe there is a MAC guru out there than can explain us how to make something easy installable out of this executable.
Macintosh:prj mr$ uname -a
Darwin macmini.pbxnsip.com 9.2.1 Darwin Kernel Version 9.2.1: Tue Feb 5 23:08:45 PST 2008; root:xnu-1228.4.20~1/RELEASE_I386 i386
Compilation seems to run smoothly. Only the clock behavior seems to be a little bit different than Linux, we could not figure out how to differentiate between real-time and CPU clock. Anyway, maybe that is just hair-splitting. So keep an eye on that, especially when the NTP client decides to change the time.
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Hi there, I have a CS410 unit and a couple Snom phones, a 320 and a 360. I have been having an issue with calls dropping if I am not talking on the call. You may wonder why I wouldn't be talking but I often do conference calls or phone seminars so the phone will be on speakerphone and in most cases my microphone will be muted. It seems to happen after about 5 minutes. This has been going on for a while with the 360 but normally the 320 worked fine. I had that problem today with the 360 again so I dialed into the call with the 320 and it was doing it too. After it did it a few times at about 5 minute intervals I tested and just before the call got to about 5 minutes I picked up the handset, unmuted the line and made a noise. I then put it back on speaker and the call went fine. If I did this before about the 5 minute mark the call would go fine but if I forgot it would disconnect again after 5 minutes.
I have the newest software release on the CS410 and the newest firmware on the Snom phones so I am not sure what the problem is.
Any help would be greatly appreciated.
It seems that there is a problem with the phone's ability to keep the connection alive during mute. From the PBX perspective the phone is "dead" and that is why the PBX hangs up. What version of the phone are you on? Maybe check release notes of firmware updates if there is anything on this.
Dirty workaround is to change the global settings for one-way audio timeout. The parameter has the name timeout_connected and you can see the current value in the pbx.xml file.
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Does PBXNSIP support DID faxing via a network fax software to email? If so what fax software is supported and what email client?
I am sorry if this sounds like a very elementary questions, just new to voip.. We currently have all our fax lines as analog lines that come in to the fax machines. We have a t1 for our incoming and outgoing traffic, is there a better way to fax incoming and outgoing than all these analog lines?
The beauty about SIP is that one vendor does not have to offer everything
. Check out www.faxback.com, they have a great solution and I heared they are running pbxnsip in their office! -
If we call them via hunt group, the phone reboots.
Ouch. What version of the phone?
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Unfortunately, a couple of my Customers do a large volume of calls at night (the majority of their calls over 1000), so I suppose there will be problems. Those Customers individually do not reach the 10,000 call mark per day, however.
So is this feature going to be "fixed" in version 3.0?
No. But you can define what "night" is. You can choose a different time zone and then assign the time zone per extension. That will fool the PBX and make it send out the reports at a different time.
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If it does cause detrimental effects to increase the number beyond 1000, like it takes up more of the server's memory until it is finally generated at midnight each night, I would think you already have the tools you need to set it up so that the server emails the CDR (and clears the memory that it was occupying) every time it reaches 1000 in a calendar day, and then goes until it reaches 1000 again and emails that list, and so on, and repeats the process for however many thousand calls per day until it reaches midnight. If memory is a problem, wouldn't that be a sensible solution?
Memory does not scare me in this case. It is just that the data collection of moer than 1000 CDR (lets say, 40000) takes a while during which the PBX would not process and new incoming calls. Now think about that call center which is working practically only at night, there we would have a problem.
If everybody is sleeping at this time, there is no problem.
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We have encountered another problem with call recording. If the Admin email address is entered in the logging page for CPU reports, and someone in a different domain on the same server sets their domain to record all outgoing calls, the Admin email account is notified that there is a recording available in addition to the party that set the recording up originally. We are using versions 2.1.12.2489 (Linux) and 2.1.14.2498 (Linux). Is there a way to disable this for the Admin and still receive the CPU reports for the server?
In 2.1 you'll have to do this with a email filter on your favourite email client... In 3.0 we introduce flags that control which emails are sent to the admin.
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There is nothing reg. on the f-key on the SNOM phone.
Have tried with factory re-set; Did not help.... (??)
Okay, did you:
- create a button profile with the park key assigned (e.g. key number 7 assigned as park with parameter 650 for park orbit 650)
- assign it to a specific extension (registration tab of the extension)
- Make sure that the MAC of the phone is associated with the extension (registration tab of the extension)
- reboot the device
Then you should see in the phone the fkeys set to button (at least the button number 7).
- create a button profile with the park key assigned (e.g. key number 7 assigned as park with parameter 650 for park orbit 650)
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I read that before and did as instructed.
I think the misunderstanding is that the PBX would authenticate an incoming Ethernet packet based on the MAC address. It does not do that - and it would be actually difficult if the request was send e.g. over the Internet. MAC addresses are kept in the packet only in the "LAN" (whatever that is). As soon as a router comes into the game, the MAC has not much meaning any more.
The MAC is only used for the automatic provisioning of devices. Even there the MAC on the Ethernet packet plays no role; whatever MAC the user agents indicates in the provisioning request is used for generating the provisioning data.
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The only line it is keeping alive is the line to my snom 370. The Siemens are not registering and the PBX is not yet capable of referring phone calls to the other extentions or mail boxes.
Line or registration? And who is keeping it alive? The phone or the PBX? I don't 100 % get it...
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I have tried this, but can not make it work.
When using "private lines" the LED do not light when used, and I can not place people on hold, just by pressing the line.
What do you have setup on the phones for fkeys? Maybe there is still something from the last plug and play operation. In doubt, just iron the phone (factory reset) and give it another shot.
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. Check out
DND problems
in snom Phones
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You you see DND set on the PBX??? That would be very strange. Does the phone accidentially send a call to the PBX and set it to DND? You should be able to see than in the call log. Also, the PBX can send an email when the DND status changes.