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Originate calls with HT503 FXO to PSTN via asterisk AMI

I am using HT503 to automate calls to PSTN. I am not sure what settings to use for this to work in India. I am able to make calls from one sip to another, so that's not an issue. I have also successfully connected FXO of ht503 to asterisk(Registered).

This is the output I see everytime I try to make a call:

> Event: Hangup

> Privilege: call,all

> Channel: SIP/amit-0000003a

> ChannelState: 6

> ChannelStateDesc: Up

> CallerIDNum: amit

> CallerIDName: Amit

> ConnectedLineNum: <unknown>

> ConnectedLineName: <unknown>

> Language: en

> AccountCode:

> Context: phones

> Exten: 995XXXX124

> Priority: 2

> Uniqueid: 1522158641.88

> Linkedid: 1522158641.88

> Cause: 127

> Cause-txt: Interworking, unspecified

Every call ends in 6 seconds, after getting answered by ht503.

My dial tone is set to:


and I've set it to wait for Dial Tone in FXO settings.

You will be helping me configure this to allow me to make outgoing calls from asterisk AMI to PSTN.

Beceriler: Asterisk PBX, VoIP

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İşveren Hakkında:
( 0 değerlendirme ) India

Proje NO: #16574424



I have worked with different fxo devices from digisol, synway, gxw, etc. Let's discuss and finish this project.

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