SIP carrier IP authentication + dialcode help needed

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SIP carrier IP authentication + dialcode help needed

Postby rickytrix » Wed Jun 20, 2018 9:48 am

Hello, today we got a new SIP provider that doesn't have user/password registration and I didn't come across that kind of carrier registration before. I googled it up and created a new carrier based on the info I found by googling. It shows the carrier listed in "sip show peers" but it doesn't show it in "sip show registry". And I couldn't make a correct dialplan entry, so if anyone can help I would be extremely thankful.

We need to call Croatia, and the provider gave us info that we should send the numbers in this format 385 xx xxxxxx. I tried configuring the dialplan but it ends up with the message "That is not a valid extension...". We used this extension for calling germany from the siptraffic carrier and it worked.

exten => _00000000149.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00000000149.,2,Dial(SIP/${EXTEN:4}@siptraffic,55,tTo)
exten => _00000000149.,3,Hangup

exten => _00000149.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00000149.,2,Dial(SIP/${EXTEN:1}@siptraffic,55,tTo)
exten => _00000149.,3,Hangup

So basically my question is: Is the carrier registered even though it doesn't show it in "sip show registry", and what would be the correct dialplan for calling a number of this format 385 xx xxxxxx. Thank you in advance, and below is the account entry and the dialplan that I tried to use.
________________________________
[Name]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
host=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
qualify=yes
insecure=very
canreinvite=no
________________________________
Global String: SIPName= SIP/Name
________________________________

exten => _385NXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _385NXXXXXXX,2,Dial(${SIPVoipInvite}/${EXTEN:1},,tTo)
exten => _385NXXXXXXX,3,Hangup
rickytrix
 
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Joined: Tue Jun 12, 2018 6:20 am

Re: SIP carrier IP authentication + dialcode help needed

Postby teleinx » Wed Jun 20, 2018 10:24 pm

When using IP auth you are not actually registered. You are sending a sip signal (INVITE) to the carrier with your vici's public STATIC IP that you provided to the carrier already and they should have it whitelisted it on their end. And you're sending the call to the IP they provided you. If there is no active call there is no communication between the 2 peers.

Your dial plan should conform to whatever your carrier asks you to send them. I.E: countrycode+areacode+number or +countrycode+areacode+number or Tech prefix plus any of the combination of the other 2 examples.
VoIP carrier spesilizing in vici dial.
Outboud VoIP Termination | Inboud VoIP Origination | Carrier Data Services
Website: http://www.teleinx.com
skype: teleinx inc
teleinx
 
Posts: 53
Joined: Fri Apr 08, 2016 7:44 pm
Location: Miami, FL


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