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Telekom
Wer kann uns helfen hier die richtigen Daten einzutragen?
Die Datei /opt/GS5/misc/freeswitch/scripts/ini/gateways.ini enthält die Konfiguration mit der Sie Ihre Telefonanlage z.B. mit einem externen SIP-Provider anschliessen können. Die Datei muss von Ihnen manuell angelegt werden. Nach einer Änderung müssen Sie FreeSWITCH mit sudo /etc/init.d/freeswitch reload reloaden.
/opt/GS5/misc/freeswitch/scripts/ini/gateways.ini
[gateway1]
profile = gemeinschaft
username = ?????
password = ?????
proxy = ??????
expire-seconds = 600
register = true
extension = {sip_to_user}
caller-id-in-from = true
##routes.ini: /opt/GS5/misc/freeswitch/scripts/ini/routes.ini
[outbound]
^%+49(%d+)$ , 0%1
<gateway description="" identifier="gateway1" inbound="true" name="Telekom 1" outbound="true" technology="sip">
<gateway_settings count="6">
<setting class_type="String" description="" name="auth_source" value="sip_gateway"/>
<setting class_type="String" description="" name="auth_pattern" value="^gateway1$"/>
<setting class_type="String" description="" name="domain" value="tel.t-online.de"/>
<setting class_type="String" description="" name="username" value="XXXXXXXXX"/>
<setting class_type="String" description="" name="password" value="ZZZZZZZZZ"/>
<setting class_type="String" description="" name="number_source" value="sip_to_user"/>
</gateway_settings>
<gateway_parameters count="10">
<parameter class_type="String" description="" name="auth-username" value="[email protected]"/>
<parameter class_type="String" description="" name="proxy" value="tel.t-online.de"/>
<parameter class_type="String" description="" name="expire-seconds" value="600"/>
<parameter class_type="String" description="" name="caller-id-in-from" value="true"/>
<parameter class_type="String" description="" name="register" value="true"/>
<parameter class_type="String" description="" name="profile" value="gemeinschaft"/>
<parameter class_type="String" description="" name="realm" value="tel.t-online.de"/>
<parameter class_type="String" description="" name="from-domain" value="tel.t-online.de"/>
<parameter class_type="String" description="" name="from-user" value="XXXXXXXXX"/>
<parameter class_type="String" description="" name="extension" value=""/>
</gateway_parameters>
<call_routes count="1">
<call_route name="default" position="1" routing_table="outbound">
<route_elements count="3">
<route_element action="match" mandatory="false" pattern="^%+49(%d+)$" position="1" replacement="0%1" var_in="destination_number" var_out="destination_number"/>
<route_element action="set" mandatory="false" pattern="" position="2" replacement="XXXXXXXXX" var_in="caller_id_number" var_out="caller_id_number"/>
<route_element action="set" mandatory="false" pattern="" position="3" replacement="XXXXXXXXX" var_in="caller_id_name" var_out="caller_id_name"/>
</route_elements>
</call_route>
</call_routes>
</gateway>
- XXXXXXXXX = Telefonnummer inkl Vorwahl (ohne +49)
- YYYYYYYYY = Die Zugangskennung (die Nummer, mit der man sich bei Telekom ins Benutzerportal einwählt)
- ZZZZZZZZZZ = Das Passwort zur Zugangskennung
Folgende Punkte sind wichtig:
- Die ausgehende Nummer muss auf die eigene Telefonnummer gesetzt werden, da die Telekom das sonst blockiert
- Die Extension muss leer sein (weiß nicht genau warum, aber sonst wird bei eingehenden Telefonate die Nummer nicht übermittelt)
- Man braucht ein Gateway für jede Nummer
Um einen Telekom SIP-TRUNK anzuschliessen, musste ich dieses Gateway erstellen:
<gateway description="Gateway für Telekom VOIP" identifier="gateway5" inbound="true" name="Telekom SIP-Trunk" outbound="true" technology="sip">
<gateway_settings count="6">
<setting class_type="String" description="" name="auth_source" value="sip_gateway"/>
<setting class_type="String" description="" name="auth_pattern" value="^gateway5$"/>
<setting class_type="String" description="" name="domain" value="sip-trunk.telekom.de"/>
<setting class_type="String" description="" name="username" value="XXXXXXXXXXXXX"/>
<setting class_type="String" description="" name="password" value="ZZZZZZZZZZZZZ"/>
<setting class_type="String" description="" name="number_source" value="sip_to_user"/>
</gateway_settings>
<gateway_parameters count="12">
<parameter class_type="String" description="" name="auth-username" value="YYYYYYYYYYYYY"/>
<parameter class_type="String" description="" name="proxy" value="reg.sip-trunk.telekom.de"/>
<parameter class_type="String" description="" name="expire-seconds" value="900"/>
<parameter class_type="String" description="" name="caller-id-in-from" value="true"/>
<parameter class_type="String" description="" name="register" value="true"/>
<parameter class_type="String" description="" name="profile" value="gemeinschaft"/>
<parameter class_type="String" description="" name="realm" value="sip-trunk.telekom.de"/>
<parameter class_type="String" description="" name="from-domain" value="sip-trunk.telekom.de"/>
<parameter class_type="String" description="" name="from-user" value="XXXXXXXXXXXXX"/>
<parameter class_type="String" description="" name="extension" value=""/>
<parameter class_type="String" description="" name="to-user" value="XXXXXXXXXXXXX"/>
<parameter class_type="String" description="" name="register-transport" value="tcp"/>
</gateway_parameters>
<call_routes count="1">
<call_route name="default" position="1" routing_table="outbound">
<route_elements count="3">
<route_element action="set" mandatory="false" pattern="" position="1" replacement="wwwwwwwwwwwww" var_in="caller_id_number" var_out="caller_id_number"/>
<route_element action="match" mandatory="false" pattern="^%+49(%d+)$" position="2" replacement="0%1" var_in="destination_number" var_out="destination_number"/>
<route_element action="set" mandatory="false" pattern="" position="3" replacement="wwwwwwwwwwwww" var_in="caller_id_name" var_out="caller_id_name"/>
</route_elements>
</call_route>
</call_routes>
</gateway>
- wwwwwwwww = Telefonnummer inkl Vorwahl (ohne +49)
- XXXXXXXXX = Telefonnummer inkl Vorwahl (mit +49)
- YYYYYYYYY = Der Telefonie-Benutzername
- ZZZZZZZZZZ = Das Telefonie-Passwort
Important: The Tables of Content are generated. Any change will be overridden on the next update.
For more information: GitHub Wikifier
Proxysetting not working at all ... we could not change the Proxy by itself
☰ 4. Routing
- Rufnummern
- Routing-Grundlagen
- Feature-Codes
- Least-cost-routing
- Prerouting
- SIP-Provider 0. Antilo 0. PBX-Network 0. QSC 0. Sipgate 0. Telekom 0. dus.net 0. HLkomm 0. easybell
- Mediagateways 0. Berofix 0. Patton Smartnode