netpro78
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Posts posted by netpro78
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https://www.inteliquent.com Currently supports STIR/SHAKEN. I recently attended one of their webinars on the service, and the coming requirements. They currently have some larger carriers like Comcast, and T-Mobile utilizing their service. My takeaway is that many carriers such as Inteliquent will have a hosted service where they can sign the calls for you (however they will only provide the highest attestation for their own numbers), the problem comes about if you are doing least cost routing, then the only real solution is to apply the certificate to the PBX, and sign all of your own calls.
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According to the FCC anyone selling services to an end user will be required to implement STIR/SHAKEN. In my opinion this is the majority of your customers that operate in the US that have multiple domains.
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Is there any plan to add support for signing outbound calls since it will soon be a FCC requirement?
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I think both are good ideas. TrueCNAM's API supports learning of false positives. You would just need to add that to your integration. I do also believe that the local address book should always take precedence.
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I recently had my first false positive spam score. I figured I could just add the number to the address book as a whitelisted number, and it should not spam score it. Unfortunately it still classifies it as spam. Is there another way to whitelist a number, or could this be added in future versions?
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Since there is now a drop down in the PBX for SugarCRM, I am interested in any documentation as to how to set it up, and versions that are supported.
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My trunks are all hardcoded to IPs, and they are working fine after 62.0 however my phones that started to have issues after upgrading to 62.0 are all using DNS if that could be part of the issue.
Also I saw some comments that 62.0 would have options to set the dynamic blacklist at either the IP, or the Port level. I do not see that option, as well as I believe a related feature was a ratelimit on the emails that it generates, and I do not see those options either.
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I upgraded to 62.0 to pickup the fixes where the PBX would accidentally blacklist trunk IPs, however after the upgrade it seems to be blacklisting phones that have never had an issue registering in the past (and their configs have not changed). If I whitelist their IPs, they can successfully register. In some cases I noticed that the phone has successfully registered, however the IP was then blacklisted after the fact when subscribing to a BLF.
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I am glad you pointed that option out. I was completely confused by "pretend to be busy" playing a recording. My mindset is that is if a phone is actually busy, and the voicemail is enabled, it would go directly there, and if the mailbox was disabled, it would send a 486 back. The message confused me since I cannot think of a situation where it would normally play it. Honestly I think the following 4 Spam options would be the most logical, and cover any need:
-Send to Voicemail
-Send Busy (sends 486, or 500 depending on trunk setting)
-Redirect (and give a field to redirect to)
-Ask for name
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VLC would run on some type of computer. I am looking for a dedicated piece of hardware (like an ATA)
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You have it when it hits the huntgroup (you can redirect directly to a mailbox), but a direct dial call to an extension has no way to send a spam call directly to voicemail.
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Are there any plans to address the options above that do the same thing? TrueCNAM even recommends that one of the options would send the call directly to voicemail.
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Thanks for the solicitation, however my customer is in the audio industry, and wants their own live feed also available as hold music. They do not want to waste/maintain a PC on site just for this purpose, and would rather have a device that takes their audio output, and converts it to a RTP stream for the phone system
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I believe you are misunderstanding. I do not want a file converter, I want a physical piece of hardware that will take a 600 ohm analog audio line input, and convert it to a Unicast RTP stream that is compatible with the PBX. Think of an ATA, however instead of an FXS port, it would have an audio input.
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After upgrading to 61.1 I have started getting notifications of IPs being blacklisted for unsuccessful authentication attempts. The IPs are from my trunks, and when I look at the attached capture, it is an Invite, and not an authentication attempt.
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Does anyone have any recommendations for a dedicated piece of hardware to create an RTP stream to the PBX to be used for music on hold? I do not want to dedicate a computer just for streaming.
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I was wondering if it is possible to have the PBX confirm voicemail deletions (such as pressing a second key to confirm), or having a way to recover deleted messages within X number of hours after deletion.
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For some reason it did not work when I copy pasted everything above, however:
from-trunk:from acd-ani ext-ani trunk-prefix dom-ani trunk-ani
Worked just fine
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Prior to upgrading to 61.1 when a call was forwarded, or a cell phone was included on the incoming call, the recipient would receive the ANI of the original caller. After the upgrade the ANI that is being received is from the extension that is doing the forwarding. I have tried all sorts of custom trunk headers, and in the past using "Based on Incoming Call" worked properly, now I cannot seem to find a combination to replicate this lost feature.
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With the latest version of your website going live, it appears that you have accidentally deleted images that the PBX uses in the emails it generates for example: https://vodia.com/img/welcomeheader.jpg
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Good catch, this specific machine was still on the Aug 27 build.
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In the testing of the new feature of "assign to the user that first opens the message" on win64 61.1 the setting keeps turning itself off even after it is turned on, and save is pressed
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Now that SPAM scoring seems to be working in my test environment, I am testing the various options for the "Handling of suspected SPAM" calls field.
the options of "Reject Call", and "Pretend to be busy" seem to do the exact same thing. They both say we are sorry, but this extension does not accept anonymous calls. The problem with this is that they both clue the caller in that some sort of ANI filtering is going on since the caller is probably aware that they are not sending anonymous. I would suggest that pretend to be busy sends the call directly to Voicemail, and reject the call sends a reorder tone to the caller (500, or 486). The other two options seem to work as you would expect.
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It seems to have fixed the issue in my test environment.
Video Conferencing
in New Features/Versions
Posted
So many phones (like Polycom and Bria) support video and back in the PBXnSIP days it somewhat worked. It seems like just being able to clean up the video support to work at a basic level so that we were not forced to turn it off on the customer's brand new phones that include video so that they will interact with the PBX properly would be a great first step.