Asterisk settings for DialTelecom

Asterisk Settings for DialTelecom

DialTelecom is an alternative telephone operator in Slovakia, that offers VoIP and regular voice services. This article shows the configuration details on how to get asterisk to call through DialTelecom.


First, edit sip.conf and add the following lines:


register => YOURUSERNAME:[email protected]     ;(the IP address is the DialTelecom Gateway)

type=friend   ;(if you want to allow DialTelecom to call into your PBX)
;type=peer    ;(if you want only outgoing calls to DialTelecom)

;   ; (the IP address for this is, the hostname I made up and entered into my /etc/hosts file)      ; this is the valid DNS record

user=phone ; (I needed this to make it work)
; (You can also enter a context, but it works without defining any)
insecure=very  ;(To allow registered hosts to call without re-authenticating)
canreinvite = no


Now, after editing the sip.conf file, it is necessary to edit the extensions.conf file, to allow calls onto through DialTelecom. In this example i will present the solution that I used. (It is specific for Slovakia, but you'll get the idea)

; For calling SlovakTelecom phone lines in Bratislava
exten => _70XXXXXXXXX                   ,1,Dial(SIP/${EXTEN:[email protected],,tThT)
exten => _70XXXXXXXXX                   ,2,congestion()
exten => _70XXXXXXXXX                   ,102,busy()

; For calling mobile phone numbers (Orange and T-Mobile)
exten => _7090XXXXXXX                   ,1,Dial(SIP/${EXTEN:[email protected],,tThH)
exten => _7090XXXXXXX                   ,2,congestion()
exten => _7090XXXXXXX                   ,102,busy()

exten => _7091XXXXXXX                   ,1,Dial(SIP/${EXTEN:[email protected],,tThH)
exten => _7091XXXXXXX                   ,2,congestion()
exten => _7091XXXXXXX                   ,102,busy()

Created by: trentin, Last modification: Thu 04 of Nov, 2010 (06:34 UTC) by admin
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