Vodia PBX
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Posts posted by Vodia PBX
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Please add call-id variable into call recording options.
Now there is no effective way to relate CDRs captured using the SOAP CDR and recordings.
The latest CDR (3.1) also include the filename of the recording ("rec").
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is there any way to add this to PNP by manually editing a file? I would prefer to use UDP so my edgemarc's can monitor the MOS score. but doing that means I have to change on a phone by phone basis, not good.
also, why doesn't the check-sync work with a snom in udp mode?
Funny thing is the RTP traffic was in deed coming outbound proxy IP, BUT the phone is on a different domain and the calls rejected on the snom. i think this setting might be bad or wrong.. there isnt a reason the phone should give me that error message.
In 3.1, we included another setting file that you can edit manually in the html directory: snom_3xx_custom.xml. There you can set whatever setting you like. It is loaded last, so it will overwrite the provisioned settings. Check the snom_3xx.xml file in the generated directory, then you get the idea.
I guess snom does not trust UDP very much. Every "idiot" in the network can send UDP packets, and having the phone reboot because a funny guy wants to boot all phones is not very productive work.
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I have been testing moving a domain from 1 pbxnsip system to another, and I am running into a problem in 3.1.1.3110 win32. When I create the domain on the new machine from the tar file, all extensions that have a DID assigned to them show up in the extension list as an extension number followed by <> if you edit the extension it does show the DID. Everything seems to run fine dispite this oddity, that is until you stop, and restart the service. Upon the restart, the logfile shows that the database sanity check deletes all of the accounts that have DIDs associated with them. I have found a workaround by editing the extension, cutting the DID out, saving the changes, and then pasting it back in, will correct the problem as long as you do all extensions with DIDs prior to restarting the service.
Creating a username that is not possible will leave the account in limbo. Using a global name twice is just an example for this. Other examples include not enough licenses.
For pure backup purposes that is fine. However when you want to create a template you must make sure that this template does not have telephone numbers in the account names.
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can you modify the dial plan at all? when they call the extension directly they have no problem so i would assume that all that needs to be done is set the dial plan to allow four digit extensions beginning with 8?
You can put the attached file into the html directory and edit it. This requires a restart of the system.
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well today... I use my fax machine and my phones start ringing, for some reason when I go to send a fax, the box shares a line with the fax machine, the local hunt group starts ringing like I am sending the call to myself, you can hear the fax on the system when answering the phone, I am not dailing my own number, why does the box answer when sending out going faxes, it has never done this before, and I have not changed a thing?
Sounds a little bit like an electrical shortcut? Not sure if you can physically share the same line for two purposes, but I am not the big FXO expert.
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Thanks for your reply,
We have a dedicated firewall and we have opened ports:
TCP: 5060, 80, 443
UDP: 5060, 49152-65534
As for the Personal firewall, we have none on the system. We do have McAfee Enterprise 8.5i, we'll uninstalled it to check.
We have also disabled TCP/IP filtering on the server.
Anything else we may check for...
I would also open TCP 5061 (TLS port).
We have assembled a checklist at http://wiki.pbxnsip.com/index.php/One-way_Audio, anything from there? Maybe something "stupid" as a IP address conflict.
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I upgraded to the latest available version, got the "try loopback" option under the trunks in the dial plan. My problem is that I can't get accounts in different domains to register. I have tried to add a new trunk to the second domain still no luck. The says the account can't be found?
Does the Request-URI (http://wiki.pbxnsip.com/index.php/Request-URI) host name match exactly the domain name? Is "localhost" still in the game?
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In the PNP how do I turn this value to OFF?
Filter packets from Registrar [filter_registrar]
I dont know if this is a firmware (7.3) issue or not. BUT when I set the snom transport = udp
I start getting 403 Use Proxy from the snom phone because of this setting. Which doesn't make sense because the proxy i sending the RTP.
Also when I use udp, the check-sync button doesnt work.... is this a tls feature only?
Well this setting is not touched by the PnP mechanism. The meaning is that the phone does not accept traffic from any other source than the PBX (which makes sense). But you should be able to change this setting if you like. Check the generated directory, there you should not see this setting.
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here is the dialplan:
dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1" dialplan.applyToUserSend="1" dialplan.applyToUserDial="1" dialplan.applyToCallListDial="0" dialplan.applyToDirectoryDial="0">
<digitmap dialplan.digitmap="[2-7]xx|8[2-7]xx|[2-9]11|1xxxxxxxxxx|011x.|*x." dialplan.digitmap.timeOut="3|3|3|3|3|3"/>
<routing>
<server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="5060"/>
<emergency dialplan.routing.emergency.1.value="911" dialplan.routing.emergency.1.server.1="1"/>
</routing>
</dialplan>
That looks like what I would expect. Looking at your original post it becomes clear why she cannot dial 8121: There is no pattern for that. The point is that extensions must start with 2-7 for the Polycom dialplan. Otherwise the above dialplan is not usable.
Is it an option to move the 1xx extensions to another location?
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i am using a Polycom 650. i have about 80 phones (mostly Polycom 320's) registered through PNP...
the four files that the system is showing in the generated folder are
1) Polycom_Sip.xml
2)Polycom_Phone.xml
3)Polycom_master.xml
4) Polycom_Addrbook.xml
is the dial plan included in one of these?
It is in polycom_sip.xml. Search for "dialplan.digitmap", there you should see the dial plan that your phone has. You should see something like "[2-7]xx|8[2-7]xx|[2-9]11|1xxxxxxxxxx|011x.|*x.".
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is there any config i can make to the dial plan to take this problem away?
besides in this case the employee is not beginning with a 1 they are beginning with an 8 as they are transferring to a mailbox...they are entering 8 followed by the three digit extension beginning with a 1!
BTW what phones are you using? Did you chech what dialplan the PBX sends to the phone in the generated directory?
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when the receptionist (using a polycom650) picks up her phone to make an outbound call and has entered the first 7 digits of the number (for example) and then a incoming call comes in, it automatically picks up the call as she has the handset in her hand and has a "dial tone"
is there anyway to set the phone/system to allow her to continue dialing the number even though a call is coming in?
What firmware are you using on the phone? That sounds like a bug in the phone to me.
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are you using PNP settings? there is a place to set the admin password. It works fine for us.
pbxnsip: what are these setting for--
Provisioning Parameters
Authentication User:
Authentication Password:
Just updated http://wiki.pbxnsip.com/index.php/Domain_S...ning_Parameters.
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yea this has been a bug for a while now.
Ops, ops ops. The problem could be reproduced... For those who would like to verify that problem is fixed, please manually update to http://pbxnsip.com/protect/pbxctrl-3.1.1.3117.exe.
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hmmmmmm.
Are we talking about the same thing???
I want to have the security function on the http access to the phones, som customer do not f*** up the settings in it.
If I need to have every phone in my hand doing this, I will cost me moeny.
It has not been an issue before, and therefore I am wondering why it suddenly is!
Further more, on the SNOM360 the phone is say "HTTP PW NOT SET", and that is a problem in it self, as the customers think something is wrong.
What I am wondering, is just, have something changed OR what is wrong??
I can not be true that this function hav been removed from one ver. to a other??
I just confirmed that the HTTP password for the phone is also set. Check your "generated" directory, there you should see that the password is also provisioned. The snom firmware is a little bit picky about this; maybe that's the problem.
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If I need to set it up on the phone, the whole idea is lost.
The idea with all this, ifcourse, that I do not need to set-up every phone
You can still choose the old modes. But none of them is really safe, that is why the default is now the challenging mode. The long-term goal is certificate based, but for that all phones must come with certificates - that will take years.
And setting up a password is still easier than setting up the whole configuration.
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in this case the user does not want to have to hit enter OR dial 9!
even in enviornments that use 9 for outbound calls i have never seen a system that requires that for internal transfers.
it is very possible that this is an issue with the dial plan on the phone but mu question is how do i get around it?
Well, as said before if you want to use the PnP dialplan on the phone you must use accounts and extensions starting with the digits 4-7. Starting with "1" means that the number will be a 11-digit telephone number in NANP area.
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Normally the system automatically set-up the HTTP PW/access to the web interface, but after upgrading to ver. 3.1
It is no longer possible for us to set up HTTP access on the phones; What I am doing wrong??
Check the (updated) http://wiki.pbxnsip.com/index.php/Snom. Probably the phone is not able to answer the HTTP challenge, maybe you can change the password policy for the provisioning or set up a password on the phone.
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When setting up the log file name under the loggin settings
if you use log$txt it will create the log file based on todays date, but i also read that it will delete any old files
is there somethign else i can put in there that will create a new log file daily based on the date and it will also leave the old log file?
It should delete old files after a few days. The exact duration is stored in the global setting called "log_keep", the default is three days.
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I called into my system and the IVCR audio was very choppy,
we restarted the server and now runs fine
any clues what might of caused it?
friday we were runnign SIPP (stress tester), could it be we left it runnign and never shut it down properly?
Well, I would assume that the choppy audio is related to the stress test. Maybe some of the calls did not disconnect properly and the bandwidth is gone with dangling calls. Wireshark would show what is going on.
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Firewall?
It could be the personal firewall on the server... Maybe some local firewall on the server kicking in and stopping the networking on the process. I don't believe and external firewall would do that.
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hope you had a nice Christmas....any update on this issue??
Not yet - still eating Christmas cookies...
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is it possible to maybe program a key on the phone to do that feuture? if not then the grandstream phones have one over the linksys as far this is option
For the company making the phone software that should be possible. But I am not aware that the PBX can make that feature available.
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i have a client who who works with extensions beginning with 1XX and 2XX. when the receptionist tries to tranfer directly to voicemail for the users beginning with 2 IE 8201 she has no problem but when she tries to transfer to users begininng with 1 it never works...i beleive that after the third digit it is automatically doing the dialing ie she enters 8121 it only recognises 812 so it says invalid extension...i would imagine this may have to do with the PNP plan which is set to three digit extensions but then why is there only a problem with extensions beginning with 1??
The question here is if this is a problem with the dial plan on the phone or a problem on the PBX. From what you write above, it sounds like a problem with the phone. What you can do is choose that the user has to press enter; alternatively the user must choose extension numbers that start with 4-7. This what most "good old" TDM PBX did as well. The alternative would have been that the user must enter "9" to get a dial tone, which requires one more key press and messes up the address book dialling. Therefore, for the PnP we chose "1" for "dial a NANP" number (10 digits).
Using the enter key has also one big advantage compared to the "get dial tone" method: You can edit the key, especially the last digit. And you have no problem dialling international numbers.
Telling pbxnsip how many concurrent calls a trunk can handle
in Trunk Setup
Posted
You can also specify which line can be only invound and which one can be only outbound. "co1:i" would be only inbound, "co2:o" would be only outbound.