Vernon

In this particular event i think it's better to restrict the device rather than the PBX since most don't support these types of actions by design. I did have a chance to try to test out the restrictive dial plan to not allow for extension/star code dialing but it doesn't have any effect because internal calls do not reach the dial plan for them to take effect. Since you're using Snom's they have an expression dial plan string page: http://wiki.snom.com/Features/Dial_Plan/Regular_Expressions The last one dictates on how to block dialing, since you only want outbound, you can substitute it with 1 instead of 9 as in the example. This way you can just send out all calls as 1xxx...etc. Since you're not stripping any digits, even if they were to dial an internal extension with a 1(xxx) it won't go anywhere since 1xxx is not a valid extension, same with starcodes. The example string from the help page: |^[^9]([0-9]{0,})$|sip:blocked\@\d

Hello, Was testing some of the features for voicemails and while this is not critical or an issue per say, i felt like it should be brought up. So after listening to a voicemail you have various options, option 5 gives you the details of the calls. Before the new version it gave you the following information: Date of voicemail Time of voicemail The call From However with the new version the structure has changed: Date of voicemail Time of voicemail The call from The call To* Perhaps i haven't tested this in the correct environment it was designed for, but it seems redundant to let the user know that the voicemail left in their mailbox was left for their number. As an analogy imagine your personal cellphone voice tree announcing to you that "Voicemail 123 has been left for your number 456" Is there a way to modify this functionality or perhaps tweak it in some way.

I'll attach an image to help better explain it. For the LDAP i tried with a Snom 745 and 725 ( 8.9.3.60 ). The domain address book LDAP works perfectly fine on these phones and picks up entries as it should, but when editing an address book from the user level and reprovisioning a phone no extra contacts appear. I've changed the setting to personal address book only and didn't see any new entries appear. I could try with a Yealink phone model and see if there's any difference

Thanks for your reply, Would you be able to clarify on Point 2? What does it mean to end that option from being selected? Selected from where? I refreshed the page with both settings and i haven't seen any difference. 4. What i meant here is that adding entries to the address book only stays within the user level address book. If you were to provision a phone it would pick up contacts from the domain address book which doesn't contain any user level entries. Though i see under Extensions -> Address Book, there is a setting for the following options: Default, Domain+personal, Domain only, Personal only. I've provisioned a phone with all variations of this setting and didn't notice any changes on the phone. P.S What is the default Address Book setting and where is it set from?

Hello, I've been testing some of the features of the User level in the PBX and noticed some bugs and have a couple of questions. Q: Question B: Bug? 1.Q: When adding an address book entry in the Web Attendant console, this entry doesn't seem to reflect in the domain/user address book. Is this entry stored only on the User Level's page? 2.Q: The third symbol of the address book doesn't indicate what it does. First is delete entry, second is to edit the entry, third, i'm assuming, is to block/allow the contact? 3.B: This might be related to the 2nd question. When adding a contact and setting the contact on the "Black List", after saving the entry and refreshing the page it will turn into a Regular contact. 4. Q: Will the address book functionality of the user interface be expanded to allow for more control over the system? For example it would be nice to let users manage their address book without having to give them full PBX access or maintaining ticket queues to add/edit entries. This was all tested on the latest Vodia IO, reading the past patch notes i wasn't able to see any notes addressing these issues/questions. Thank you

Hello, I have a quick question. When creating domains and editing it the voicemail transcription option will be enabled by default. When navigating to the E-mail -> Notification and setting the default option to "Off", it will still create new domains with the setting enabled. Which setting takes priority? Right now it looks like you still have to edit individual domains to disable the transcription each time a domain is created

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.

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.

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.

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.

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.

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

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.

Yep

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.

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.

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?

Thank you!

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.

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?

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.

Hello, Was wondering if you could provide any insights on how to resolve no audio on calls made via the user portal. I tried following all the necessary steps, i have a domain certificate, using chrome, and i have logged in via HTTPS. I'm able to make the call go out and i'm able to put the call on hold with the MoH kicking in, but no audio between users. Currently using v59.8 on Windows. Any insights would be helpful, thank you.

Yeah i meant GAPS with the Snom PBX not with the phones . I had the same scenario of having to add the MAC's into the GAPS beforehand. Thank you for taking the time to respond, i guess you and i had similar experiences with it. I'll be on the lookout for when the special feature hits.

Has anyone had luck integrating the GAPS with Snom? I was trying to integrate it via the Phones -> Settings -> Provisioning Server Redirection section with the username/password for the account. Tried inputting a MAC for one of the grandstream phones on a test extension but it didn't seem to create any records on GAPS. I was testing this on the 56.0 release of Vodia.

Hello, Thanks for replying. I do see agents sometimes appearing during the general recorded calls section but that seems to occur only if someone dials an agent (extension) directly. In regards to the Agent Group it never seems to show up even though that information is technically present during the call log. See below for example. Agent Group Call Log: Time From To Agent Waiting Time Ring Time Talk Time (Hold Time) 9/21 1:53P "Anonymous" <sip:anonymous@domain.domain.com> "Center" <sip:Telephone#@domain.domain.com> "Jaime" <sip:224@domain.domain.com> 00:15 00:06 02:20 00:44 Agent Group Recorded Calls: Start From To Agent Play 9/21/2016, 2:10:20 PM Anonymous (anonymous) Center (Telephone#) ## Sorry for the poor formatting i simply copied and edited the data. ## indicates no data. So there are multiple agent groups and all the agents are logged into them. The call flow is designed so that all calls go through the agent groups. I'm looking for guidance on how to have the Recorded Call section properly show the Agent/Extension involved in the call as the PBX clearly has this data but is not displaying it in other sections. Thank you