Can't able to load carrier details
Hi, i am a newbie here and trying to figure out how to use this dialer. I need to know what should write in carrier id, carrier name, global string and on the starting of account entry fields. I tried everything like my username, my name etc.. but nothing is working. My carrier details are not getting registered in dialer as its showing registration error, timed out, cannot connect in asterisk realtime logs. but same SIP settings are getting registered and working fine on eyebeam. So, basically i am unable to figure it out whats wrong in it. Also, a snapshot of the fields i don't know about what to write is also attached. Any help on this issue will be highly appreciated.
My account Entry:
[abc]
nat=yes
disallow=all
allow=g729
allow=ulaw
type=friend
username=5002
secret=64610425
host=198.154.109.41
dtmfmode=rfc2833
context=trunkinbound
insecure=very
port=5060
canreinvite=no
fromuser=5002
My Dialplan for US:
exten => _05X.,1,AGI
exten => _05X.,2,Dial(SIP/${EXTEN:2}@198.154.109.41,,tTor)
exten => _05X.,3,Hangup
Replies (3)
RE: Can't able to load carrier details
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Added by Demian Biscocho over 9 years ago
Looks like you're using an older version of GOautodial. Please download and install the latest version (3.3). You can easily add a carrier via our web based wizards. You can also use our JustGoVoIP VoIP service which is integrated into the new dashboard. You can see your VoIP credits/balance in near realtime.
RE: Can't able to load carrier details
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Added by zap virus over 9 years ago
Thanks for the reply but i need to run this dialer only as i am much familiar with this one. Also i just can't figure out what entries should i write in certain fields(see the image for more details. Any help regarding that issue? thanks in advance.
RE: Can't able to load carrier details
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Added by Demian Biscocho over 9 years ago
Sorry. We don't support versions older than 3.0 anymore. Please upgrade. The newer versions are easier to use. You can also check the available documentation in our Wiki to get you started quickly.
This might help though: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf.