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Everything posted by John

  1. 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?
  2. 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
  3. Thank you very much, it worked. Note for those with multiple domains: the extension should be in the form extension@domain Regards, John
  4. 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
  5. 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 ...
  6. 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).
  7. 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.
  8. - 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
  9. John

    E-mail templates

    Found it in the translation.txt, thank you very much.
  10. 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
  11. 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?
  12. 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.
  13. 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
  14. 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).
  15. 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
  16. Hello, I have the following setup for a customer: Auto Attendant with a DID assigned to it and dtmf options configured. The caller hears the welcome message and presses the key to connect to the department he wishes to. The Auto Attendant redirects the incoming calls to 3 agent groups depending on what key the caller pressed. Each agent group has 2 agents, all of them logged in and available. The "Number of agents added per stage:" is set to 2 (basically we want the agent groups to act as hunt groups but with music on hold and messages for the callers in queue). 2 of the 3 agents groups work as they should. If both agents are available, their phones will ring simultaneously. In the third agent group however all calls are always sent to one agent only. The second agent's phone never rings. For some reason the pbx consider the extension as unavailable. From the logfile: [7] 20150414152709: Call 17178: [ACD 9101] Next stage in 9101 has 1 agents available, 483 selected The other agent (484) though is available (I also made sure that dnd is disabled both in the pbx and in the phone). Both users use Yealink T20P phones with the same firmware - Even if I register more user agents for the second user, his extension won't ring. If I set another extension as the second agent and call through the Auto Attendant, both agents will ring. If I assing a DID directly to the agent group and call it, both agents will ring (agent 484 too). If I set "Allow multiple ACD calls on agent - even if busy:" to yes both agents will ring. So, what are the possible reasons that make the PBX (v 5.2.5) to inccorrectly consider this specific extension as unavailable? Regards, John
  17. We have a similar request that requires a way for the caller to display different caller-ids: A secretary works for a group of companies. She makes calls on behalf of 5-6 companies and she wants to change her caller-id on outbound calls based on which company represents each time. Can we do this somehow? We offer a hosted service and the pbx server can only have one sip trunk with the sip proxy. We are the phone company as well and we can allow a customer to display multiple caller id's. Thanks, John
  18. John


    Thank you for your reply Christian, If I understand it correctly, only we have this problem. Where is the html directory located? I could't find it (it isn't under the /usr/local/snomONE directory"). Did you mean the working directory? What files should I delete? Do you have any suggestions for the playback issues in Internet Explorer or Chrome? We don't necessarily have to use Firefox but it was the only browser that worked.
  19. John


    Hello, after upgrading to 5.2.5, the Recorded Calls page in Firefox doesn't display the list of calls anymore. For instance, I recorded a conference call. The file does exist in the file system. This is the Recorded Calls page in Firefox: At the same time Internet explorer and Chrome list the recording but they can neither play nor save the audio file: Firefox was the only browser were playback and file-saving were working without issues. Can you please check it with priority and provide as with a custom build if need be (customers have already complained). Our OS is Debian. As for Internet Explorer, the following may be helful to you: if I right click on the audio player and choose "inspect element" this is what I see: This is not the case though. I tried html5 audio playback in other pages and it works. I only have issues with the pbx. Regards, John
  20. Hello, It is an ad-hoc conference room and I was moderator. Anyway, consider it solved. After restarting the service last night, the star codes work again. John
  21. Hello and happy new year to everyone, Debian, v5.2.4 hosted here. Some of the star codes used during conference calls don't work for us. Specifically, the working ones are: *1 for muting the moderator's device *2 for unmuting the moderator's device while the non-working ones are: *7 for muting the other participants *8 fro unmuting the other participants and *9 for hanging up the conference call We tried first with a Yealink device and then with a mobile phone (gsm call, not through a soft phone). The conference room type is ad-hoc. The DTMF tones appear in the log as below. They just don't work: [6] 14:07:07.926 APP: Received DTMF *, call type extcall [6] 14:07:07.926 APP: Received DTMF *, call type conference [6] 14:07:08.406 APP: Received DTMF 7, call type extcall [6] 14:07:08.406 APP: Received DTMF 7, call type conference Please check it and if needed, fix it in the next version. Thanks, John
  22. Hello, I need your guidance for the implementation of the following scenario: A customer (who has his own non-Vodia PBX) wants a SIP Trunk with the PBX for in- and outbound communication. For the inbound calls the requirement is that all calls that receive a busy (486) tone from the remote system must be automatically forwarded to another/ fallback number. I at first tried a SIP Gateway trunk and played around with several settings in the trunk settings page without success. The redirection to the fall back number (I use the “Redirect destination when all lines are busy:” for this) does not work. I am now thinking of a SIP registration trunk approach. An extension on the PBX will be registered to the remote PBX. I will send all incoming calls to that extension and whenever the remote PBX returns a busy signal the call will be automatically forwarded to the fallback number (set in the “Include following extensions when this extension is being called:” field) . A drawback in this is that the customer has ten incoming numbers and in case of busy the calls to each number should be forwarded to a different number. Thus, ten registration trunks are needed and I am not sure if he would accept such a solution. My other concern is the caller ID. Whatever I have tried, the PBX does not forward the caller's caller ID but the domain or the extension's ANI. Does anyone have another suggestion? Thank very much in advance, John
  23. John


    One more bug we discovered in 5.2.4: In the field "Extensions that may jump in or out (* for all, includes non-agents too):" in an agent group the only acceptable values are either the * or nothing. When I enter anything else (specific entension numbers), the system returns the following error: In this case, extension 187 does exist. Is there a work-around? I need the functionality provided by this field urgently to test a custom call distribution requested by a customer.
  24. I can confirm the issue (v 5.2.4). Please fix it in the next release.
  25. John


    Minor bug report: after upgrading to 5.2.4 the system sends out the daily PBX Resources e-mail twice with a minute delay between the two mesages. The second e-mail retains the text and the layout but the graphs contain no data.
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