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The ongoing Firewall Saga! (Victims - I mean volunteers - wanted!)

Well, no-one said that it ate their system over the weekend, so I'm going to stick this into the online module repository today or tomorrow some time, so I can expose it for a bit more testing. Anyone...

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I have a problem with loss of audio briefly during a conversation

Thank you so much for your help. I will be onsite tomorrow and their IT guy is supposed to be there as well, I'll show him this response and see what he can do to help get this issue straightened out....

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Reload failed because retrieve_conf encountered an error: 137

Thank you! hostname -v shows correct name. Or you mean something another?

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FreePBX/ CUCM 8.6 SIP Trunking - Calls not routing

I may be too late in replying and you might have found this out yourself. I guess the issue is when you try calling from CUCM -> FreePBX. The Free PBX sends a 401 Unauthorized. This is because of...

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Same issue as yours

Continuing the discussion from Trunk context problems: Majren: Thanks for the help SkykingOH. I appreciate it. Hey Majren, I am having the same issue as yours, i have created an Inbound route for a...

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The ongoing Firewall Saga! (Victims - I mean volunteers - wanted!)

No feedback, but just to say that when it becomes available for Debian I'll be happy to give it a go and feedback.

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CID Superfecta - GooglePlaces

I can reproduce this: Add Google places to a superfeacta scheme. (click yes next to it on the list) Click settings for Google Places Click Yes in the box that comes up. Click Save in that box As a...

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Caller ID Superfecta - Who Called - broken

This correction doesn't seem to have merged correctly into release/13.0 - both the live to systems superfecta version 13.0.3.2 and release/13.0 have the old (non-https) code, so currently systems have...

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Design of bootstrap and module to asterisk linkage

I'm trying to mkake sure that I understand the design of bootstrap, and the design framework for modules as best I can before I make any other comments or suggestions for improvements. I have probably...

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PJSIP - multiple devices per extension - missed call

Hi,thanks for your help. Yes, Rob, that is what I am looking for. This seems to be the same way, as it worked with ring groups with the chan_sip driver, isn't it? Because, that always worked quite...

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Caller ID Superfecta - Who Called - broken

For some reason those links fail when I click them, but I take your point. Weird, must be a failure at my end. However, it is not https in: github.com...

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FreePBX Distro Iptables

I recognize this is an old post, but it shows up in google, so I'm going to answer it the way I did it. Also, there are a number of posts in this forum regarding this topic: SkykingOH: The SIP...

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Design of bootstrap and module to asterisk linkage

Continuing the discussion from Design of bootstrap and module to asterisk linkage: tm1000: ntadmin: if there is a design doc out thereThat said. Please post what/how you think it all works here....

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Which git repository

This is probably obvious to those in the know, but as I've tried to understand this project and get involved, I've just realised that there are two repositories. For example, for superfecta, I could...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

Were you being sarcastic on your last post? Sorry, I am really in doubt. I can give access to my FreePBX box that has this problem. If anyone wants to log in and take a look, be my guest. What I need...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

mbello: Were you being sarcastic on your last post? Sorry, I am really in doubt. No. This issue needs to be brought up to Asterisk. There's nothing FreePBX can do.

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Design of bootstrap and module to asterisk linkage

That is correct......

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Which git repository

POSSA Superfecta hasn't been migrated into FreePBX fully. It will at some point. So sometimes it's slightly behind. Doesn't matter which one you work on

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FPBX13 RC, G722 and 1-way audio - even to voicemail

Well, maybe someone from FreePBX side would be interested in digging a bit, even if to get it fixed on Asterisk? If some Asterisk expert can take a look at it and open a ticket it will have better...

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Weird Call Flow question

I know what I'm about to ask can be fixed by using queues but we have an extremely stubborn customer so here goes; We have a customer that has a ring group that has itself set as the destination. I...

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