Devam Ediyor

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:

f1=400@-10,c=0/0;

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

Daha fazlasını gör: grandstream ht503 manual, grandstream ht503 fxo configuration, freepbx fxo gateway, grandstream freepbx setup, grandstream ht503 setup, grandstream fxo 3cx, fxo gateway asterisk, grandstream ht503 asterisk, asterisk ami originate wait, asterisk ami originate, forward pstn calls pstn asterisk, forward calls pstn asterisk, asterisk ami originate autoanswer, asterisk ami gnu, asterisk ami php

İşveren Hakkında:
( 0 değerlendirme ) India

Proje NO: #16574424

Seçilen:

xuniltech

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

%selectedBids___i_period_sub_7% gün içinde 5000%project_currencyDetails_sign_sub_9% %project_currencyDetails_code_sub_10%
(22 Değerlendirme)
5.3