Troy L. Scott
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- Joined
- Aug 10, 2016
- Messages
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Afternoon all,
I have attempted to set up the SMS dictation to text app in on the incrediblePBX FreePBX.
Asterisk (Ver. 13.7.2) on a raspberryPI 3.
I have had some problems with the gvoice authentication but I am able to dial out on my two trunks one is a google voice motif and the other is one that is provisioned with the simon_telephonics sip trunk.
it seems my extension 701 I dial from a android phone using CisSimple app works but I get a hung channel and the 767 internal ext is still active.
Channel Location State Application(Data)
SIP/701-00000004 767@from-internal:42 Up System(gvoice -e 2016.t23456@g
1 active channel
1 active call
4 calls processed
I have issued the following asterick command to free up this hung channel but it still shows up as problematic
Astericks CLI > channel request hangup SIP/701-00000004
Astericks CLI> channel redirect SIP/701-00000004 703
I am able to use the sip extension for calls but it still shows as connected to 767@from-internal:42 Up
I can restart astericks it is not a production system and no one will care. I just would like to learn how to forces a call to release for future reference.
Thanks in advance for anyone providing their input..... ?Still in the learning curve?
Respectfully,
T.L. Scott
I have attempted to set up the SMS dictation to text app in on the incrediblePBX FreePBX.
Asterisk (Ver. 13.7.2) on a raspberryPI 3.
I have had some problems with the gvoice authentication but I am able to dial out on my two trunks one is a google voice motif and the other is one that is provisioned with the simon_telephonics sip trunk.
it seems my extension 701 I dial from a android phone using CisSimple app works but I get a hung channel and the 767 internal ext is still active.
Channel Location State Application(Data)
SIP/701-00000004 767@from-internal:42 Up System(gvoice -e 2016.t23456@g
1 active channel
1 active call
4 calls processed
I have issued the following asterick command to free up this hung channel but it still shows up as problematic
Astericks CLI > channel request hangup SIP/701-00000004
Astericks CLI> channel redirect SIP/701-00000004 703
I am able to use the sip extension for calls but it still shows as connected to 767@from-internal:42 Up
I can restart astericks it is not a production system and no one will care. I just would like to learn how to forces a call to release for future reference.
Thanks in advance for anyone providing their input..... ?Still in the learning curve?
Respectfully,
T.L. Scott