John

This isn't the case for us, the base installation was 5.1.3 which already included password policy (and I think medium is the default value). Is there any other way we can find more info about the incident apart from the log in /var/log/snomONE?

Hello, as you know we operate the hosted edition. Two accounts in the default domain pbx.company.com (localhost) gotten hacked. We have never used these accounts, we didn't even knew the sip passwords until after the incident Are the sip passwords for the extensions in the default domain the same after each installation or are they generated randomly? Because it is unlikely the intruders acquired them through a brute force attack since we have set an ip to be blocked for a week after three unsuccesful registration attempts. Thanks John

Thank you very much, it worked. Note for those with multiple domains: the extension should be in the form extension@domain Regards, John

Hello, I would appriciate a fast reply on that: a customer bought a Soundastation IP 6000 and wants me to set it up for him. I played with it remotely for a few minutes but I haven't succeded registering it. Does anyone have experience configuring this phone? Which settings I need to adjust to get it registered? Thank you very much in advance, John

Still no luck with the log file I will keep trying. (I am thinking of installing the 32 bit version of Vodia tsp). Meanwhile, have you (or anyone else) know of another program? I can't find anything ...

Hello, My mistake. By 1 I meant Enabled (after enabling it and clicking Add you will see why I got confused). From a security perspective sounds right (and not just for Yealink phones. For instance, I have seen Polycom phones with the same issues). But further and more thorough testing is required. I mainly have experience with T2X series and I am not sure if these settings are available in all Yealink phone models. Also note that account.1.sip_trust_ctr wasn't available in previous firmware versions (If I remember correctly it was introduced in version X.72.0.30). By the way: Yealink T20(P) is EoL and according to our suppliers Yealink T22P(P) as well (although still listed as a current model in Yealink's web site).

Yep, these aren't calls. Someone is scanning you using sip vicious. This is how I got rid of these calls for good: 1. Update the Firmware, 2. Go to Features --> General Information and set Allow IP Call to Disabled, 3. Use the Yealink Configuration Generator Tool and find the option account.1.sip_trust_ctrl. Select it, Set value to 1, save the configuration file and import it to the phone. If you have more accounts on the phone, you need to do this for all the accounts. Hope it helps.

- Vodia Tapi Service Provider v2.04 64 bit - Outlook 2013 (Office 365 - I don't use Lync) - Windows 8.1 64 bit When I use the Windows Dialer I can make calls without issues. But when I use Outlook only the first call is successful. After the call has ended Outlook still "thinks" that the call is in progress: Κατάσταση: κλήση ... means Status: call ... Λήξη κλήσης means End Call If I choose to End the Call the Status changes to Terminating Call ... but nothing happens. I have to restart Outlook to be able to make a new call. I tried to provide you with the logfile by editing the registry but regardless the save location I choose, the logfile isn't created (it goes without saying that I have admin rights to the computer). As for the option Auto Originate Outbound Calls in the TAPI Service Provider seems to not work. Whether I select it or not the call starts automatically (I use a yealink phone). Apart from the above, have you ever used some other program that works with the PBX? I need to suggest something to a prospect so I either have to make the TAPI Service provider to work or to suggest something else. I tried teletrigger but this is buggy too. I would appreciate any other suggestion, free or paid. Regards, John

Found it in the translation.txt, thank you very much.

Hello, we are making some changes to the e-mail templates. Through the templates page on the PBX we were able to apply any customizations we wanted to. The only exception is the text "The best SIP-PBX since 2006 now from Vodia Networks". Please tell me where I can find and change this text. Regards, John

Hello, thank you for your reply. We tried your suggestion and it works. However, this is not a suitable solution for the hosted edition since the customized file can serve one customer only (the url for the sip-trunk is hardcoded). We won't be able to provide the same service to any more customers. Any suggestions to that?

Hi all, we are currently experimenting with the webRTC feature. Everything works fine in Chrome, Firefox requests permission to use the microphone and then nothing happens and as for Internet Explorer, clicking the "Make Call" button does nothing. Anyway, we have the following customer case: the customer wants to add the talk button to their company website and receive incoming calls from their visitors. However, because of the keypad below the "Make Call" button the dialer interface is confusing. Some visitors think that they have to call a number which they do not know and as a result they don't proceed with the call. Question 1: How can we customize the dialer interface? We primarilly want to hide the keypad and leave only the "Make Call" button. And it would have been nice to change the logo to our own (well, this seems easy, I assume that we only have to change the url sources for the images in the code provided by the pbx). Question 2: Where is the web server running? Question 3: I noticed that when the call button is pressed there is a dynamic javascript object that gets instantiated and gets fetched from the server (/usr_callbutton.htm). Is this file located somewhere in the file system and is this the one we have to edit? (I am aware that my questions may sound naive. I am not the web developer myself, I just want some initial information to pass to him). Regards, John PS. Dynamically altering the resulting web page (using javascript to hide elements) is not an option for us.

Two suggestions for improving this feature: 1. Add a way to import multiple numbers at once. Currently it is possible to add the numbers-to-be-called only one by one (I don't want to mess up with the file system). The PBX already has this functionality in other areas, you could extend it here as well. 2. what is the purpose of the prompt? After the agent has dialed the ACD star code the PBX should immediately call a number from the list. Or at least make the prompt optional. Regards, John

Hi, yes it does. But it doens't work even if the call forward is set to another extension in the same domain (where the dial plan isn't involved).

Call flow scenario: Auto Attendant --> Hunt Group --> Extension. If the extension has enabled the call forward for all calls (either to another extension or to an external number) the call won't go through. The relevant exempt from the log file: [7] 20150416123500: Hunt Group 500: Moving to next stage [7] 20150416123500: Hunt 500: started 0 calls [7] 20150416123500: Hunt Group 500: Moving to next stage [7] 20150416123500: Hunt 500: started 0 calls [7] 20150416123500: Hunt Group 500: Moving to next stage [7] 20150416123500: Hunt 500: started 0 calls [7] 20150416123500: Hunt Group 500: Moving to next stage [6] 20150416123500: Hunt: Last stage, no destination [5] 20150416123500: SIP Tx udp:XXX.XX.X.XXX:5060: BYE sip:XXX.XX.X.XXX:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP XXX.XX.X.XXX:5060;branch=z9hG4bK-20769709c3828e26cf75ead238ebaeb1;rport Route: <sip:XXX.XX.X.XXX;lr;nat=yes;did=946.92b3d243> From: <sip:+3021XXXXXXXX@anonymous.invalid:5060;user=phone>;tag=67575b5e4a To: <sip:+3069XX@XXX.XX.X.XXX;user=phone>;tag=552F8237-194402E0-0AB0143D Call-ID: 552F8237-00728DD2@hiqpcu-lyk-236 CSeq: 24836 BYE Max-Forwards: 70 Contact: <sip:+3021XXXXXXX@XXX.XX.X.XXX:5060;transport=udp> Content-Length: 0 The BYE is sent by the PBX to the SIP Proxy (I have a call trace I can sent you by e-mail). When I remove the call forward, the extension in the hunt group rings. Please check it. Regards, John