Vernon

I'm not sure how it would look like, but you can try the FROM-Header settings inside of a ring group if you can route calls through there. Auto-attendants might have this feature too depending on your PBX version. There should be an option that shows only the group name and should strip/remove the number/name.

That would be useful for those that wish to use the best and most stable v68 before switching over to v69

In the drop down when choosing the setting you wish to change, you can type the first few letters of the setting name and hopefully it will pop up the one you want. This works well with settings that have unique names, but the real trouble comes with overlapping setting names such as "Send..." They are usually grouped by tab/page so that would explain why it wasn't done alphabetically

I recommend you change the setting for one extension at least once before you start bulk editing. The reason behind it is that you would see the proper/correct value. In this case the Announcement mode settings are either false/true. Or blank if it's never been modified based on my testing. False is Anonymous Announcement True is Name Announcement

Are you customizing the login page? I saw in other posts that having a custom login page will usually start giving you problems if you don't make some adjustments to make it work.

I'm not sure if that's app related as i don't use it, but normally if you have audio issues internally (extension to extension), then the likely culprit is something on the network. Either a firewall, or bad WiFi device dropping packets...etc You can try to register two softphones on a desktop, or try to use the web client of the PBX and see if you can re-create the error in different environments. One inside the office where you know it happens, one from a home office with different networking equipment. But if this ONLY happens on the iOS app then that would be harder to troubleshoot.

I think the easiest way would probably be through service flags, if the number you are forwarding to is static each time you could create a manual service flag for this condition but your trouble will be managing it remotely. The next easiest option is to register a softphone client, if possible, and to dial the manual service flag to trigger this forwarding rule. Doing the redirects on the trunk level is something only an admin should have access to, try to utilize the other system features. If the forwarding number changes from time to time, you can still use the softphone to redirect to your desired destination by have the service flag redirected to the soft phone extension and then using *71 call forward all to set the destination. Just a few ideas, hope that helps. I just realized this is a super old post. Sorry! Hope this helps for future viewers!

If you have v 68.0.32 then you should have this feature in the mailbox tab if it's a regular extension. I didn't see you mention using this option so i attached an image of how it should look like

It would definitely be a nice feature to have. The only alternative i can see is using a multi-tenant setup, but that hampers the web client experience since not everything is under one domain. You could make the calls work inter-domain but at this point it might just be making the issue more complicated than it's worth.

Try the mailbox tab instead of the settings in "General"

Is this fax extension setup for FAX as the mailbox method? Otherwise you probably have the PBX voicemail duration set to 5 minutes

Auto-attendants and Ring Groups should have the same service flag functionality. I only mentioned a separate ring group so that it would be a check before going to regular call flow. So it will look like RingGroup 1 (Service Flag Checker): only final stage send to Auto-Attendant -> Auto-Attendant (Main Incoming): Press 1, Press 2...etc time out go to Ring Group (Main Ring Group) -> Main Ring Group follows your stage settings Basically you're just adding an extra layer on top of your call flow that checks your service flags.

Is the incoming number always the same? You can definitely use the expression list to match specific numbers to be routed to your desired destinations. Depending on your version you could also do it based on the address book. For example you can create a hunt group that does a service flag check for address book entry. e.g Create Hunt Group with final stage that follows your desired call flow. Create address book entry and name the incoming numbers with easy to recognize category, lets say "VIP". Go into your hunt group and at the bottom of redirection select "Caller in addressbook", service flag account is the category VIP, night service number is the new destination you want this number to go to. The only problem with this approach is obviously the level of scale you want to deploy, if it's one or two numbers. easy to manage. If you are looking route calls based on area codes...etc, expression list is the better way to go. There should be a couple posts about expression lists, you should search for them if you want to get a better understanding of how to deploy it.

I have added the ability to only allow specified domains to a user. This way if you have a user/reseller with many domains, they have a simple portal to find and use.

I have been developing this system for several years, and I think some of the Vodia administrators here would also find it useful in their businesses. If this is not allowed, please let me know and I will remove it. If you are interested in getting a demo, simply DM me. Unified Vodia Server Management System is a comprehensive tool designed for administrators and technical support teams managing one or multiple Vodia PBX servers. It streamlines the management process by allowing owners/administrators to allow specific privileges to support agents as well as the required tools to efficiently find tenants and support them, therefore enhancing security, and improving efficiency in handling client requests and server maintenance. Key Features Multi-Version Compatibility: Seamlessly connects with Vodia server versions 57 through 68. Support for version 69 coming soon. Unified Interface: Manage all server versions through a single, user-friendly dashboard. Role-Based Access Control: Limit technical support agents to tenant admin functionalities, excluding server admin privileges or global/superadmin settings. Restricted Admin Access: Provide support technicians with admin rights to each tenant domain, without access to advanced PBX features. Read-Only Access to Dial Plans and Trunks: Allows viewing of dial plans and trunks without the ability to modify them (can be allowed in Vodia user control.) Domain-Specific CDR Access: View Call Detail Records for each domain individually. Enhanced Search Capabilities: Easily locate servers and tenants by DID, domain name, or alias by simply entering the domain name, number or alias in the filter field. Comprehensive Client Notes: Create and view notes for each client. Notes are immutable and timestamped, accessible to all users, but only modifiable by admins. Server Management: Admins can add new servers, test connections, and manage user access. Benefits Improved Efficiency: Quickly locate tenants or PBX systems using advanced search options. Enhanced Security: Role-based access controls prevent unauthorized modifications to critical settings. Streamlined Training: Simplifies the training process for new technical support agents. Centralized Management: Manage multiple systems effortlessly from one central location. Detailed Record Keeping: Maintain comprehensive, unalterable notes for future reference and accountability. Ideal for: Businesses with one or multiple Vodia PBX deployed in the cloud or customer premises. Administrators seek to maintain control while delegating support tasks. Teams require a secure, efficient, and user-friendly management tool.

Maybe official support can give you a better idea. You can try on a different browser in case you have something that interacts with the icons. If that's the account receiving the call then the file should be visible to you on any admin level.

I test it by routing calls through a switch where the trunk is a gateway, but my trouble is not the provider settings but the unknown changes. It makes a big difference for multi-tenant PBX's because fixing one trunk is easy. Fixing 200 is quite an undertaking. As an example, you have your front door, you have your one key, you live on V68 Avenue Road, and everything just works without any issues or changes. You decide one day to upgrade your home lifestyle and go down to V69 Street, you take with you to your new house, yourself, your door, and your key that you know works for this door. But to your dismay your key now no longer works when the door is installed on V69 Street. The text view is also a lot nicer now, before all the text values were all over the place, but with V69 the format automatically changes that the most important values are up top and the generics are at the bottom. This is a nice addition. I will definitely review the logs a little more closely the second time around and see if there's an easy/fast change that can be made to accommodate the transition from v68 to v69

If you have an audio recording already on the AA, you should be able to first play it, then on the media player press the three dots, that will let you download it.

Hello, I've been testing V69 and when i created the trunk configuration that worked in V68, it surprisingly fails on V69. So far the biggest contributing factor that i can find is that when the trunk is setup with "Rewrite Global Numbers" to Use + Symbol at beginning. V69 interprets this to modify inbound numbers as well. So lets say i have an inbound DID: 7854561234. This DID is on V68, i update to V69 with my V68 trunk configuration, and the calls will begin to fail consistently. From what i can see (the logs are very sparse for some reason), the PBX is expecting +7854561234, irrelevant if the country code is set or not, it only changes how the DID is presented on the interface. When i manually add an extra did as 0117854561234, inbound will begin to work. As soon as i downgrade the PBX back to V68, no issues whatsoever. I went over the release notes trying to see if a new setting or option was introduced into the trunking, but nothing sticks out. There appears to be undocumented changes (format is different of trunks when viewing in text view). Based on my findings perhaps the best course of action is to revert the logic change, and then introduce a new setting/feature in the trunk that edits/rewrites INBOUND numbers. It would definitely help other users with similar setups to cut down on spending support time trying to figure out what went wrong during the update.

I don't think the PBX will allow you to loop the calls indefinitely with this scenario. There needs to be an exit strategy. In the older versions there used to be a setting hidden in the .xml that was called max hops, maybe the new version has it as max loops. Take a look and maybe increase this value to be higher so the PBX will tolerate more hops/loops. If you are using v68 and you want callers to "loop", you should be using the agent group instead of ring group. Then inside the agent group create your announcement messages, which i assume that's what the AA's are for.

There used to be a pretty good option in the older versions where from the auto-attendant you could dictate what extensions could or could not be called. But this feature has been removed in favor of the far more restrictive Groups feature in recent versions. You might be able to do it under group settings but i recommend testing it. You could try playing around with the IVR node To based routing match list, but you might have to read some official documentation or engage the support team for that.

You could always go via the manual route. Call the participants and then transfer them into the conference extension.

I think you'll first need to have two calls on your web application before you can use the Conference feature. The conference button from my testing doesn't function like a conference feature from a SIP phone, it will instead merge/transfer all calls into an available conference room on your domain. You should be able to try it out internally, call one registered test extension, put 1st call on hold, call 2nd registered test extension, and finally press the conference button. All calls should now be redirected into the available conference room. The calls don't have to be internal, but those are just my suggestions for testing verifications.

Have you checked the Group permissions? I think the later versions have created "Group" permissions and you might have to explicitly allow extensions/users to play recordings and use the web/phone app.

Thank you for the link to the website, i was using something extremely similar. I was hoping to also find out all the other expression mentioned in the wiki such as !T! or !E! This was the format i used, might be a bit ugly but it worked on testing. !(+1205|+1251|1205|1201|205|201)!100!f! !(+1480|+1420|1480|1420|480|420)!101!f! ....etc As long as the string is properly put into the FROM based routing match list, it should flow into the assigned destinations (e.g 100,101...etc) Based on the documentation the expressions are read left to right so i'm assuming the FROM is also read that way so shouldn't run into a situation where if these digits exist in the middle of a number (239-205-1234). It would be awesome if there was a built in expression checker, something similar to what used to exist in the dial plan, but those are the old Snom days. Thank you again for the suggestions and i hope this post can help shed some light for future searchers.