Vodia PBX
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How does pbxnsip know the MAC of the phone if you use the above URL with the Provisioning admin login?
Good point. In the 3.3 build (3.3.1.3177) we had to change this. You must use the username that should be provisioned; but you may use the password of the provisioning settings for the domain. This has the benefit that the user may change his or her password and still reboot the phone and get the provisioning data.
Important: Make sure that you have the domain provisioning data setup. There were versions that generated a default password, which is easy to guess and with that an outside attacker can easily get the password of the domain's accounts.
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it would be nice to see an option for "press 4 to send to voicemail but monitor the call" so you can listen to the person leave the VM. even cooler would be to allow the caller to break into the call to resume the voice conversation.
LOL. That could be "scary cool" - then the caller hears the voicemail annoucement he probably adds a comment nobody is supposed to hear!
Maybe during that time the caller should be muted ...
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how do you get cisco to provision? it looks like they only like tftp.
Provisioning Cisco phones remains a tricky topic. You make one little mistake, then the whole config gets rejected. We have templates for the 7960 and the 7961 built-in; however did not work on them over the past few versions.
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Yes I can pickup the call dialing *87+extension number, but my specific situation is that not allways whom needs to pickup the call knows what extension is ringing.
Did you try the "Explicitly specify pickup preference" in the extension setting?
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I did try that and it extends the number of CDR's in the Status / Call Log page.
It did not extend the CDR's in the personal webpages.
The list size is always 25, you cannot change that right now.
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Are the message wav files actually deleted from PBXNSIP or are they just not able to be accessed?
No they are really deleted immediately.
We have another issue with the shared mailboxes that is related to that problem. When a message arrives at a shared mailbox, the PBX better copies that (with the link to the file) into all other mailboxes - and then when one of the recipients deletes it, all other messages must also be deleted. We better this this problem as well, catching several birds with one stone...
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We are using 3.3.0.3152 (Win32) and I waited 45 (our time out) seconds and there wasn't any additional prompt played.
Hmm. Maybe try 3.3.1.3173 instead?
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A caller comming into the Auto Attendant dials a non-existing extension number.
The Auto Attendant responds with "This extension number does not exist".
Could this prompt be amended to say "This extension number does not exist, please try another."
Some callers just hang up because they are unaware that it is possible to try another extension.
Okay.
However, after some time the caller should hear the prompt to enter the number?
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If a message is deleted is it possible to retrieve it?
If not, when might this feature be incorporated into PBXNSIP?
Yes, this is on the list for the remake of the mailbox. So far if you choose to delete it, it is deleted immediately.
Best way to avoid the loss is to send it out as email. Then you can decide what to do with it in your email client.
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Any update?
Is it fixed on 3.3.1? download link?
So far the state is that the PBX does record, although the direction sometimes gets screwed up. 3.3 is still not out, still some issues that we don't want to release.
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Is there a * code or a way to pull a call back to your extension after it has been forked to a cell phone without hanging up and returning a call to the other party? I can see how this might be a potential problem with someone jacking your call if you are out of the office. Perhaps to help reduce that risk, there can be configurable option so people to have to enter their voicemail PIN to use this option.
If you call that extension again the callback will be canceled.
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I have my cellphone number entered in my extension account.
When I call in on my cellphone I am given a choice of Voice Mail, Auto Attendant or to make a call out.
If I make a call out I need to hang up a call back in again to place additional calls.
Is it possible to press some digit code like ### to release the call and deliver a new dial tone to me?
Yes, ## should work in version 3.3.
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Can the Calling Card be made to use Call Back function?
Yes, there is a callback function like that. In the calling card account, check the "callback" button.
There is one catch with the hotel room: In many hotels, there is no DID number for the hotel room. This is a interesting fact, because there are more DID numbers in this world slash in the USA than IPv4 addresses (10^10 > 2^32). For that you would probably need that recording asking that nice lady at the front desk to put the call through to the room that you have just recorded. Not sure if you can trick all hotels with that, but at least some of them.
The other thing that would have to be added would be entering DTMF digits after the call got connected. This is to get through auto attendants.
Version 4!
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When I call into our system and I am answered by our Auto Attendant if I press * or # it just sits there.
I entered the * and the # in the Direct Destinations area of the Auto Attendant Account and it still did not work.
* generally means "clean up the input". So when you press it, the PBX thinks that you know what you are doing an stops talking to you.
# is usually "enter". If there is no input, there is nothing to enter. Okay, obviously in that case the user does obviously not know what he is exactly doing so the PBX should be talking something.
If you want to really call a star code, you can do that from the mailbox main menu. There you are "logged in" as your extension and there is no authentication problem.
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There isn't any loop back in the dial plains and that would only affect the dial plan. I found it after doing some real searching and it's called "Accept Redirect" to yes instead of the default trunk setting of no.
The "Try Loopback" is a kind of hack to deal with the situation that many installations realls just have one server and don't want to add anything to the installation. Once you are running multiple servers, you need to have something external that redirects trunk calls to other domains on other servers anyway.
You need to disable Loopback Detection (admin->settings) and you must have IP addres 127.0.0.1 on that server (which is no problem if you are running IPv4).
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as of now, i have reverted back to 3.2 until 3.3.2 comes out. with 3.2, my polycom conference phones work and my snom's provision great over http (remotely) as well as internally. Although i would like to use 3.3.X because of some nice features, especially customized email messages.
3.2 is pretty stable. If you don't need the L&G this is very reasonable.
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Our pbx is attached to two Vegastream PSTN ISDN Gateways (Vega 50 6x4). The gatewas are connected to another phone system. So we have to dial a "0" for acces to the public telephone network.
After upgrading to Version 3.3.0.3165 (Win32) we have the following problem: when dialing a local phone number (for example 0 12345) we get no connection. When we include the area code (for example 0 02204 12345) everything works.
With version 3.2.0 everything was fine. We didn't change anything in the trunk settings for the dial plans. Any ideas / suggestions?
Here are excerpts from the logs:
Did you set the country code to 49? If yes, then check the dial plan again. The patterns in entries there should *not* have prefixes. You can add prefixes in the replacement, but the PBX internally puts everything into a global format, which starts with 00CCAA (CC=country code, AA=area code) or 0AA or just the local number (the way you would write a number down on paper).
For example, you could use pattern 0033*, and replacement 00033* if you want to put a zero before the number if you want to have a rule for France.
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What missing parameters are you talking about?
There are couple of places you have PnP related settings
Admin->Settings->PnP page
Domain->Settings page: Provisioning Parameters section
The snom PnP is still a little bit work in progresss, I think we have to come out with a 3.3.2 build that seems to clean all the problems with the snom PnP up (and also some smaller issues we found with Polycom and Linksys). If you can hold it for one or two more days, hopefully then we have a version that makes plug and play a positive experience.
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the wiki presents this regexp:9[911|411]
and explains:
The first pattern matches the emergency number and the service number explicitly and sends it to the local gateway.as far as I know (based on the linux grep/sed tools) the above is not correct as it will also match e.g. 914 and it should be corrected to
9(911|411)Am I correct or are there any big differences in the regexp that pbxnsip uses compared to those that most linux utils understand?
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PS: I would love to see a text box and a [test] button on the dial plan page where I could type a number and after hitting "test" I could see bellow the log lines about rules it matched and the replacement made -- it would save me a lot of time!
The example 9[911|411] uses "simple" expressions, while 9(911|411)@.* would be a ERE expression. Both are bad examples, because in version 3 there should be no more 9 prefix for outbound calls (this just creates a mess with the address book).
Thinking about a version 4 dial plan I see more the problem that customers want to define by the number prefix which trunk the call should take. I have seem long long dial plans that scare me. Maybe we can just have something like a CSV list that defines which prefix gets routed on what trunk. Just thinking aloud here.
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There is one problem I wonder if you can help me with. We try to recording all income/outgoing calls, we have problem with the file name and recording the dtfm input. According to Wiki (http://wiki.pbxnsip.com/index.php/Recording ), $i is used to indicate direction of the call (incoming i /outgoing o); and $u/$n are replaced with the calling/called party number. But it is not working in my case.
Looks like that for the incoming call, it's always use our public phone number as calling party. For example, I called from my cell phone
(770-XXX-XXXX) to our public phone (678-373-4755), The file is "16783734755-16783734755-111547-i.wav" which is not correct. It should be "1770XXXXXXX-16783734755-111547-i.wav". Also, when I called from an extension (5090) to my cell (770-XXX-XXXX), the file is "5090-917703135695-114558-i.wav" which is not correct. It should be
"5090-917703135695-114558-o.wav" . Here is my setting for the Record–
Location "$r/$d/$u-$n-$t-$i.wav" .
Out of band-DTMF is problematic as the PBX just passes it through without converting it into inband.
The direction problem is because the PBX has problems determining if this is a inbound-outbound (redirected) call or not.
We probably have to change the recording model so that we define the point where the recording takes place. For example, it would make sense to define that the recording always takes place on the trunk call leg. At the moment the model is not very clear and that leads to difficulties understanding why something is inbound or outbound.
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Is it possible to set that on 11/24 were open from 9Am till 12:30PM and on 11/25 thru 11/28 were not open at all?
i want to know if i can set this in 1 service flag, or do i need to build diff flags
maybe in hopliday i can have 11/24(09:00-12:30) 11/25 11/26 11/27
and that will set that 11/24 the service flag will be set except bwteeen the hours of 0900 and 1230
You can combile different service flags. When you list them, they are processed in a sequence until there is a "service active" (see http://wiki.pbxnsip.com/index.php/Auto_Att...t#Night_Service).
Also, the AA has the possibility to play back different annoucements for different times (IVR tab). That should make it possible to cover most cases.
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Pattern Replacements
611 19251234567
Well that should work... I saw examles for 411 and they did work fine.
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This may be the wrong forum, but I was wondering if anyone else has the same issue. All our snoms 360's and 370's are on 7.135. When we reboot the phones the vlan tag is lost, when you go into network you can see the vlan is still set to 1, but you have to clear it, then set it again and reboot. This happens to about 50% of our phones.
Are you provisioning the VLAN tags? What version of the PBX are you using?
If you do that, you'll need DHCP servers in both VLANs (VLAN 0 and whatever VLAN you are using) and the PBX needs to be present in both VLAN for the provisioning.
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isn't the failover scripts/whatever in plans for next version of windows ver of pbxnsip?
Yes. But if the way we are doing it would also work with version 3 it probably also does not hurt.
Recording
in Embedded
Posted
I believe it would be too complicated to get recording on this device. There are two things that come to my mind. First, you could mount a NFS partition and record on that partition. Second, you could use the monitoring feature of the PBX to send all calls for call recording to an external destination (as a SIP call). In both cases, you need a recording license and enough CPU power.
For serious recording operation, I would recommend to switch to a server/PC and do it there.