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System updates gets exception even after reboot

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hi @marbled: I dont have any DAHDI or Digium devices. What is wanrouter module?

I'll wait i have enough other issues :slight_smile:
Thanks leon


Vega FXO gateway - calls won't connect

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My Vega gateway will receive calls and make local calls from FreePBX. However, when I dial a long distance call, the call will not complete, and I get a recorded message from the telco that the call could not complete as dialed.

I am in the USA. In the logs, the call disconnects "call cause 16". Any ideas on how to get this working?

Thanks.

Outbound calling doesn't work

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While it might have fixed your problem, I don't think it's the solution.

There's something else going on here that we need to skull out. Changing the time condition to a set time like this shouldn't affect the way the route is processed.

Voice mail not recognizing touch tones

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I have FreePBX 14.0.1.4 installed. I have a couple sip phones connected. (for the record they are TEO phones, but that does not help much). The phones are connecting to the server ok. I can call the other phone ok. When i dial 97 to check voicemail voicemail answers but it does not respond to any button that i push (2,3,0,,#). Does anybody have any suggestions as to what to look at to figure it out?

Thanks

Vega FXO gateway - calls won't connect

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Not sure, since a SIP failure would have a SIP failure code like 416. Cause 16, from the quick perusal I did, appears to be an improperly formatted outbound phone number. Are you add a '1' that shouldn't be there, or missing a '1' that should be?

System updates gets exception even after reboot

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Hi Leon!

The actual name of the driver is Wanpipe,

wanrouter is the script we run to start, stop, etc... it...

re: https://wiki.freepbx.org/display/PC/Card+Driver-+Commands

It is a Sangoma telephony card hardware specific driver... It can actually do more than DAHDI but with FreePBX/Asterisk it's a driver we must load in addition to DAHDI...

We use it with cards such as these:

I don't think there is much risk if you don't have any DAHDI hardware but as the MTA gals/guys say "Your server, your rules"... :wink:

(At one point in time I was really into mail servers (ie MTAs)...)

Good luck and have a nice day!

Nick

Voicemail message deleted from mailbox

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Sorry to resurrect an old thread, but I have actually had this issue on four separate deployments. It happens randomly and different people. I have also reviewed the Asterisk full file and see the message being left but nothing about it being deleted.

Like gw22, when I go into the voicemail settings for extension in question, the the "Email" option is disabled and the "Delete" option is enabled.

In my cases, the users do not have access tot he UCP, so they would not have set this themselves, accidentally or otherwise.

There is definitely a bug in the system that sets this, but no rhyme or reason for it.

If anyone else has come across this and/or knows what is happening, thoughts would be greatly appreciated. It is hard to deploy this as a production-ready system when glitches like this happen, and the best we can do is shrug our shoulders and move on :frowning:

System updates gets exception even after reboot

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Hi Nick...ok thanks for the info....nope just running SIP stuff on a VM :slight_smile:
I'll watch for your yeah or nay to move along thanks

Leon


Clearing Edits From Postfix main.cf Before using Sys Admin email setup

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I am using FreePBX Distro 10.13.66-21 very successfully. I have just moved from slow as molasses static business DSL which allowed port 25 usage to higher speed cable, non business class dynamic IP, which blocks port 25.

When I first setup email for my voicemails and etc.,I didn't have the commercial System Admin module and had to add email manually editing postfix' main.cf and etcetera.

SInce port 25 is blocked, I now need to use something else and gmail fits the bill. So before I start trying the Email Setup in System Admin, should I take my main.cf and other modified files for manual postfix setup back to their default? I ask becasue it seems freepbx will over write them anyway...??

Thanks for any info!

John

Voice mail not recognizing touch tones

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Sounds like a problem with DTMF......

Yealink T4XG phones will not autoprovision over HTTPS with FreePBX 14

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Where are the logs that should be recording failures? The ssl logs (both ssl_access_log and ssl_error_log) are empty as shown above.

System updates gets exception even after reboot

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I read that but there is something very important missing...

Both the version of the Wanpipe drivers and the version of the kernel is important... They are essentially built against a kernel AFAIK and while they support being used after a kernel with very subtle differences with the help of weak-modules this has pretty big limitations...

Some combinations result in broken Wanpipe drivers because of a mismatch between the Wanpipe drivers and the kernel...

It sounds like it's only with a non-workiing Wanpipe drivers combination that things work for you without further interventions needed like restarting...

All of the Wanpipe drivers published worked for me when paired with the right kernels. Maybe not fully smoothly but as long as I was willing to have them timeout after 4-5 minutes at boot I never had to restart anything...

Good luck and have a nice day!

Nick

System updates gets exception even after reboot

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Ok...

Before the big update a few days ago, DAHDi would only start if I did a service dahdi start, then fwconsole restart. I don't know what version of dahdi & wanpipe I was running at this point, but everything was being kept up to date.

After the big update (which I assume updated the both the kernel and the dahdi & wanpipe drivers?), everything started working correctly for me. At this point my Wanpipe driver was at 7.0.20-9.sng7.

Today the wanpipe drivers were updated to version 7.0.20.13-1.sng7. Having updated these drivers, once again DAHDi will only start if I do a service dahdi start, then fwconsole restart.

Dial extensions limited by directory

Inbound calls not working


Inbound calls not working

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I found where the mistake was. I had the ports backwards between Chan_sip and PJSIP. Inbound calls are now working. Thanks all.

Twilio Inbound route calles get “the number you have dialed is not in service”

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How do I fix this, do I change the configuration on astrsik to accept anonymous sip

Detecting Original distination number on incoming forwarded call

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I have been searching for weeks on a way to route an incoming call based on the following conditions:
1. The incoming call was forwarded.
2. read the originally dialed phone number.
3. Route the incoming call to specific destination based on a match on the originally dialed number.

In short, I want to create a voicemail platform that will recieve busy/unanswered forwarded calls and route them to the proper mailbox from a single DID.

My provider is flowroute and the calls would be forwarded by AT&T Wireless.

What would it take to do this? I have tried looking into Asterisk Logs but the call information does not provide any info that would help divert a call that was forwarded. I suspect that this "detection" would need to be done on the SIP header level.

System updates gets exception even after reboot

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Wanpipe was actually broken at 7.0.20-9 with the new kernel. Since you have unsupported hardware that is acting like a digium card this might have made things better for you temporarily. Now that it works with supported hardware but broken for your unsupported it hard to say. You're basically having the same issues others are so nothing has changed and we are where we were at last week. Which is a good thing.

System updates gets exception even after reboot

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Yes. Back where I was a week ago.

There is another member on here with the same issues that I'm having, who is using a digium (supported?) card:

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