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About Vernon

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  1. You can try utilizing the CO lines for a specific trunk, then when it runs out of lines it should prioritize the busy routing you've created for that specific trunk. That way you can keep your attendant without having to use any other accounts.
  2. Hi, I'm trying to figure out how to best approach a scenario where a manager has to take over calls. Let's say Manager runs a call center and Rep A received Chatty Client B. CCB has frustrated RA to a point where heated words may be said. Manager wants to be able to grab the call from RA without having to use parks/holds. I know there's a feature for call barge in, but that would still leave RA on the line, is there a possibility of just completely taking the call over? To summarize the feature would be a forced transfer.
  3. I think he's referring to the example in the paging group under "Destination" From your Wiki: https://doc.vodia.com/paginggroups Under "Setting up the Paging Account" Destination (e.g. "40 41 5*") The example destination makes it look like it would page all extensions in the 5x range.
  4. Are there any plans to reintroduce the button templates for the SPA series phones? SPA504, 509, 525. Seems the only Cisco phones that have a template are the CP series and the SPA 112/190 (which have empty button profiles) This is on version 61.1 The provisioning still works just can't modify any of the buttons.
  5. For those looking for a possible work-around for the time being is to copy/paste the phone-MAC.xml file from the generated folder into the extensions custom phone xml. e.g snom_745_phone.xml...etc It seems like the first extension in the list gets the settings for both extensions, while the second identity phone-MAC.xml only generates for itself. It may not be pretty if it's for the whole system, but for individual extensions it should work well enough.
  6. Hello, I was just wondering if the Shared Line/ co line functionality will be making a return in the new version 62? I remember reading in one of the blog posts that the shared line will cease to exist in version 60+, but i can see from the new button list in version 61 that there is allocation for co lines. Provisioned a 320 with the buttons being co lines and accounts co1, co2, co3...etc Similar to how it's described in your wiki: https://doc.vodia.com/trunk_colines, but doesn't work like the older versions. Just to recap, in the past people would press a shared line and the button lit up on every phone to indicate the line is in use. On version 61.1 the button provisions it as a BLF that enables monitoring, but the line can't be selected so this does not have much usage. The value is provisioned as sip:co1@test@testdomain;user=phone>|*87. I'm guessing the value is provisioned in such a fashion to push for the usage of park orbits. Thank you
  7. Doesn't seem like that label template affects the 745. I've tried making it empty or even putting other unrelated information in there to see if it would affect it. Always seems to come up the same, it's account + name in the label. Is there another configuration where the 745 would be picking up the private line label? Giving the label an empty character/space simply forces that label to be empty.
  8. So far the 745 is the only one i noticed that the label actively messes with the button feature. I posted a picture just to better explain it. I'll explain the process. Provision 745 on extension. No button profile means no labels created for first time provision. Move to create button profile and put 2 private lines. Press save and all buttons except private lines show up. For testing purposes i went into the web interface and set 2 more private lines but with no label. At this point the picture will help. Basically what's happening is the private lines with label no longer show basic information such as dialed call or a parked call. In the picture i dialed a test extension, but doing the same with the provisioned private lines shows nothing but the label. This is why i don't think it's sch a problem for other phones because this functionality i've only noticed on the 745 and not any other phone.
  9. Does version 60+ have a modifiable button configuration? I noticed the xml's for the phone specific buttons have been migrated to the new button scheme (much friendlier). There is just another xml for snom_buttons and inside is just: <functionKeys> {snom-buttons} </functionKeys> I think if we would be able to modify the way the label feature works it might solve the issue for 745's. To be honest this kind of issue would probably only show up on a 745 or similar phone that has buttons on the screen. Would you be able to elaborate on the snom power mode for the button parameters? i haven't heard of this before and would like to read up on it.
  10. Hello, I noticed one small little issue with the Snom 745 and the provisioned private line. Unprovisioned the private line of the 745 simply says Line [Free] and will display caller information when calls are received. However once the button is provisioned it picks up the automatically filled extension details, thereafter blocking any future caller information on any of the buttons if programmed for private lines. Is there a way to reverse this functionality or to stop it?
  11. Hello, I was just reviewing the button list under version 60.2 and i see it's lacking the MWI (Message Waiting Indicator). Has this been shifted into a different functionality or system menu? According to: https://vodia.com/documentation/buttons the MWI should be present but i just do not see it. Also i noticed under Mailbox -> Call Cellphone/ Call Supervisor, the number of attempts have a drop down, but once you save and refresh the page the attempts go to 1 attempt. Is this just a visual bug or does it still update to the correct amount of calls? Same as above testing on Version 60.2 / Windows.
  12. Definitely has the webRTC support under the status. Quick question, the domain certificate i'm using is for all domains for example *.domain.com. Should i instead just get a temporary one specifically suited for my test domain?
  13. Thank you for your response, I'm familiar with how to resolve one way audio issues between devices. Most notorious ones are with SIP ALG, double NAT's, and firewalls. However this is a test server with no firewall rules or any complicated setups. I'm assuming though when the call is made via the user portal the RTP range that's used is the one under SIP -> Audio? I've never used WebRTC before so i'm not particularly familiar with how to troubleshoot it.
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