I have voice mail hints turned off, but still getting the MWI on voicemail app.
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Downgrade PhoneApps-Mandatory red MWI light not needed on voicemail app
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Google Voice just stopped
Interesting...well it just decided to show connected again, however only outgoing calls work. Incoming calls go to Google Voice and are not forwarded to Asterisk box. Is this change recent?
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Google Voice just stopped
I don't use GV but started seeing complaints yesterday.
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Google Voice just stopped
Hmm, yeah im not 100% sure whats the issue. I can make outgoing calls, any softphone client on a desktop or smartphone will register and make ext-ext calls no problem as well as outgoing. Just incoming doesn't work, and when I do call my GV number, it redirects to another number that then leads to my Google Voice voicemail. Not real privy on Asterisk so not even sure how to check oAuth version. I do know its 11.4
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Google Voice just stopped
oauth2 is not supported by asterisk upstream. It is a patch people add.
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Google Voice just stopped
Ahh,well if the issue was oauth2, wouldn't that prevent PBX from even connecting to GV? Its connected now, just incoming not being forwarded
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EPM Grandstream variables
I'll answer my own question, it was obvious, well should have been, ops. The values are taken from the feature codes in Freepbx.
Now to edit 44 templates with the new P values.
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Google Voice just stopped
(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)
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Google Voice just stopped
So interesting, I left myself a message and the GV service actually called in on my GV number to let me listen to the message, so the incoming call directly from GV worked.
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Google Voice just stopped
Well must have been something with Google. I did another reboot of Asterisk and now its all working again. So I guess problem solved =)
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Keep outgoing caller ID for External queue call
I'm using my flowroute SIP trunk.
There has to be a way to override it - I can override it if I call direct, so I don't see why I can't 'change' it when a call is made from the queue...
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Google Voice just stopped
Sorry, I don't really know anything more about it, except that yesterday/today's issue seems to have affected Obihai and Asterisk equally and there are a lot of inconsistent reports about how the problem was resolved.
The error message at face value seems to point to a TLS incompatibility, as though Google were enforcing a more strict TLS scheme and the client side has an older OpenSSL implementation that is not compatible. That hypothesis would line up with Google's usual push for stronger, more secure encryption, but there is no proof that this is the case.
The other thing we saw while troubleshooting it on the PIAF forum was users only being offered OAUTH2 as a login method -- user/pass ("plain") login was not being offered in the XMPP handshake.
It's not known whether people resolved the connection problem through various updates or it went away on its own (i.e. Google did something, or it was a temporary error on some of Google's XMPP servers).
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Downgrade PhoneApps-Mandatory red MWI light not needed on voicemail app
Please open a feature request to add a option to not turn on the light. This has always been the case but a bug in hints was making it so hints were not working correctly so you would not always see the light. That bug was fixed for all BLFs
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Conference Menu
Conference Pro module does this for you and also lets you have it auto create a conference room for every user for you.
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Disable In-Call Asterisk Toggle Call Recording via COS
I would like to restrict usage of the In-Call Asterisk Toggle Call Recording feature code.
I am thinking COS might be the way, but the feature code is not available for "Deny/Allow" in COS.
How do I allow it for some extensions and deny it for others?
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WEBRTC phone register but unavailable
You need to use webrtc version 13.0.25 or later. 13.0.27 is the latest and comes from the Edge Track.
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No RTP engine was found/Failed to authenticate
In the source directory I installed from, "make menuselect" shows under "resource modules" that "res_rtp_asterisk" and "res_rtp_multicast" are selected already.
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Conference Menu
Thanks
I have a workaround by using an IVR.
Pressing 1 takes them to room 1, 2 to room 2 etc. This means I only need one incoming DDI
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No RTP engine was found/Failed to authenticate
[root@ivr01 asterisk-12.8.2]# pwd
/usr/local/src/asterisk-12.8.2
[root@ivr01 asterisk-12.8.2]# find -name rtp.o
./bridges/bridge_native_rtp.o
./channels/chan_multicast_rtp.o
./main/sdp_srtp.o
./main/rtp_engine.o
./res/res_rtp_multicast.o
./res/res_rtp_asterisk.o
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No RTP engine was found/Failed to authenticate
Here's a wrinkle. I installed from 12.8.2 from source but I'm running 13.6.0 after some upgrades. Thoughts?
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