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  1. Today
  2. as for modifying to *97 seems to me to be the most appropriate solution. about the behavior of the template for Yealink should check it, maybe I'm wrong. With 2 previous versions of Yealink it worked for me, with last fw it doesn't work. Please copy me a couple of examples that you enter in "Yealink General"? I make a comparison with my template. Thanks Damiano
  3. Yesterday
  4. Hello Community we did not find a solution to cover TAPI and CTI Integration out of the box with vodia and after a long time we have been able to fully support vodia for UCPlus. All functions DND, call forwarding and integrations are supported. ucplus is a powerful unified communication software suite, which allows users a high level of integration, communication and collaboration with their phone system. Strong integration to a huge range of different Business applications systems users information about callers quickly and efficiently significantly increasing productivity. The ucplus product suite is not only cost effective but also very easy to maintain and install. UCPLus can be used as a multi-tendant for hosted solutions or in-house for a very good price. We use UCPlus for our hosted Vodia in multi-tendant operation Information can be found here https://plus-software.com/uk For all who are also on the search this is over thanks
  5. Hmm did Yealink change the behavior that the first occurrence of a setting is the one that will be used? If that is the case we will have to move the yealink-general to the bottom of the template. Or we just put it to the top and the bottom. Please make sure that you did not introduce a XML syntax error - then the whole file will not be accepted by the phone. The voicemail depends on a system setting. The "Calling own extension number goes to mailbox" (in dom_mailbox.htm) controls that. By default this should be on - if you turned it off yes then it will cause that problem. I believe in the next version we should use *97 instead - we will change the template to make sure that this is not causing the problem again.
  6. Hi, It is a race condition. What might have happened is, if you have set your "Maximum number of recordings (system wide)" value to too less, for e.g. 25000. When the system hits 25000 CDRs then it would automatically erase them as that condition sets to be true before your 30 days condition. So even before it could record for 30 days, it gets cleared up due to heavy number of call volume on your server.
  7. We had enabled call logging to go back 30 days and pull each call by extension so we could send our client an export of the csv on this. It appears that call logs only go back 12 days – are you able to confirm the locations we configure this to ensure we have not missed something?
  8. Last week
  9. hello, I normally enter the customizations for Yealink Templates Phones in the "Yealink General" window, and all phones regularly take these custom parameters. I have upgraded my Yealink phones to version 66.84.0.90 and now it doesn't work anymore (even if I enter the customizations in Yealink General, these parameters don't get them from the Yealink phone). Now to customize the Yealink template I have to edit the file "yealink_common.txt". Is this a known error? Is there a solution? ------------------------------------------------------------------------------------------------------------------------------------------------------------ I also found a bug (I think), basic Vodia sets the listening code of the voicemail = voice_mail.number.1 = {account} Then pressing the voicemail key from extension 215, for example, will dial 215, so it's an error. To fix it I tried to insert: voice_mail.number.1 = *97 The problem is that the * character doesn't take him in charge, in fact if for example imposed (only for test): voice_mail.number.1 = 333 At this point, by pressing the voicemail key, the telephone sends 333. This shows that it is currently impossible to set *97 for voicemail listening. Solution?
  10. Hi, We recently activated the TAPI integration with ConnectWise Manage for our support desk however we quickly found an issue that led us to having to disable it straight away, ISSUE: With TAPI enabled and ConnectWise open we can make calls from the desk phones however afterwe trigger a call from ConnectWise and then try make a call from the desk phone all iInbound and Outbound calls to/from that extension show Busy as soon as the call starts to ring. If we then try to trigger a call from ConnectWise the call works fine. If we close ConnectWise then the desk phone works straight away no issues. If we open ConnectWise again it remains working until we trigger a call from ConnectWise. I've mentioned ConnectWise alot however we are also talking to them to see if there are any issues they could assist with at the same time.
  11. Ringback usually means that a human is being called and that the call will connect within seconds. Generally we want to keep it that way. That is why the ACD plays that sound when an agent is being called for the current call. However there are situations where this is not desirable, e.g. when agents are available only sporadically or when it takes a long time to get an agent to answer the call, and you want to keep the caller on the phone. Yes The way this currently works is that there are no more announcements rendered into the music - a very subtle hint that the ACD is actually ringing an agent! You can actually have multiple calls ringing. For example the ACD can assign the call to a specific agent and then schedule the next agent, even while that call is still ringing. For that, you can set the "Allow multiple calls to ring agents in parallel" option to yes.
  12. I would have thought agent group vs auto attendant to distribute would be good but the issue is it needs to be super simple for the end use, the request mainly comes from small business so it's not large call flow volume. The use case here is super simple, example being. Office is closed for x-mas party etc, admin staff want an easy way (Star Code) to re-direct all incoming calls to the Domain i.e user DID's + main business lines to a specified exit point that might only be known on the day/hrs before so manual input would work best. That way they don't have to rely on the staff forwarding their phones. It would be nice for it to use the agent group type feature set as its base of operation to take advantage of call queueing with multiple exit points if desired and the press 1 to connect option for agent exit point hunting. Most times these would be mobile exit points so you don't want mobile voice mail to answer the call. But for it to work like that it would need to be simple i.e, starcode -> domain redirection menu -> input destination/s -> set on/off/clear etc. Which would intern login a virtual extension to a virtual agent group with basic settings as mentioned above. My thinkings
  13. Hi. I just wanted to clarify the Ringback tone functionality on an Agent group "Agent Group / Caller Setup / Audio / Ringback tone" When a single call is in the queue and an agent is available: Setting it to "Regular ringback" plays the ringback tone, but non of the periodic announcements (wav file 1, wav file 2, etc) - which understandably because caller one is not technically in the queue, it is in the ringback state. However when set to "no ringback tone" the single caller in the queue hears the music as well as the periodic announcements, but presumably this single caller is still in the same ringback state? My question now being...With the "Regular Ringback" setting, is there also a way for the single caller to hear the periodic messages as well while they are waiting for an agent to pick up the call the same as when "no ringback tone" is set? And also...is there a way for callers 2+ who are in the queue to hear a ringback tone instead of music? Thanks,
  14. I would not random down or upgrades without knowing what is causing the issue. Downloading a domain should be pretty safe - except when there are so many records that you might run out of memory. Maybe you'll see the pattern and then once it is reproducible we can say what the problem is/was.
  15. Earlier
  16. Yes we are using the RPS server - The pbx crashed last night when I pressed the download domain button to copy a domain. So it isn't consistent, it is intermittent and has only happened a few times. So far as I can tell, it silently drops out and I lose the admin website and all phones lose registration. It won't start with just a "service pbx start" - I need to perform a "service pbx stop" first. I can only connect via SSH and will check with "top" when it happens again (although I don't want it to happen again during business hours) I also didn't have these problems before 63.05, i am not sure if that is co-incidence?? Should I downgrade to a previous version as a preventative measure??
  17. My thinking would be around using an auto attendant that can distribute incoming calls where we can control the redirection with a star code (aka winter storm redirection). But what is the use case here?
  18. Of course if it happens when a specific action is taking place that helps a lot with identifying the problem. Also the question is what exactly happens - for example does it just stop accepting new connections (which would hint at problems with the number of sockets), does it go berserk with the CPU, or does it just silently drop out. Are you using the RPS service?
  19. Hi Team, Did a search but could not see any discussion. I don't see any doco on the "Call redirection" setting under domain settings yet. I am wanting to know if there is a hidden star code not yet published where the user can dial it to set the forward destination for the whole domain like with the *80+hunt/agentgroup option.
  20. Hi Guys, Recently I have come across an issue with the pbx service stops and the only way to get it back up is issuing a "pbx service stop" then a "pbx service start" command. It seems to happen after a change has been made to the system. Most recently it occurred when moving phones via MAC id's. It has happened 3 times now and usually in the middle of the day ippbx version 63.0.5 Linux version - Ubuntu 18.04.3 Any ideas would be much appreciated. Thanks in advance Christian
  21. IOP does not respect contact & connect headers in device's INVITE.

    remote phone is connected via site-to-site IPSec VPN. The SIP settings are set to:

    SIP replace 10.10.10.108/71.78.XX.XX

    IP route is set to 10.10.10.0/255.255.224.0/10.10.10.108 0.0.0.0/0.0.0.0/71.78.XX.XX

    The remote phone is 10.10.30.101 and the contact & connect headers tell the IOP to use the private IP of the phone, but the IOP responds back and tells the remote phone to use the public IP of 71.78.XX.XX and not the private IP of 10.10.10.108. The subnet mask used in the replace should encompass the 10.10.30.X range, but it seems to not.

    I have two VPNs set, with 3 different subnets seen at the IOP:

    10.10.10.0 local

    10.10.20.0 remote

    10.10.30.0 remote

    Running a 63.0.4 release.

     

    Do I need to have a separate entry for each VPN'ed LAN to use the local IP of the IOP?

  22. Oh what did you put into the field? Maybe there was something the backend did not understand which would explain why the email never went out...
  23. We experienced this issue also. We found that leaving this field blank was the solution. The pbx will then send the agent group email at midnight
  24. Do you have anything in the log where we can see what exactly is sent to the PBX? For example turn the logging for Call-related SIP packets on - then we should see what is coming in. Also, do you have a country code set for this domain?
  25. 1) If you have agents in an Hunt group logged in2) And if you have this feature turned on.3) And if you have one of these agents set as a BLF button on an extension ( who's not the part of that Hunt Group)4) If an incoming call falls on that HG then you will be able to pick the call from the non-agent phone via the BLF button.
  26. Hi, I've got some difficulty to configure caller ID on inbound calls with my Vodia IO. When i call the 09 82 99 97 14 with my mobile phone which has got number 06 52 23 88 87, on my extension screen i see "652238887". How can i add the 0 before the number to have the correct display on my screen ? Here you can found my configuration for the moment. If I choose "use a list of expression" my call don't work anymore because my opmerator say "This number is not assigned" If i configure like the screenshot, call works but display is not like i want Thanks for your help julien
  27. Is it like this, suppose I've a Hunt group 400, and I've registered a phone as extension 200. Now on this phone I configure a key as BLF 400. Now if someone calling at 400 on my phone I see BLF line gonna red that mean call is ringing at hunt group 400, so by just pressing that configured key I can take that call? Is it how it will work?
  28. Hi, Can anyone tell me please how can we use this feature "Enable call pickup from extension BLF" in the hunt group. I read the vodia document but did not understand complete. Thank You
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