<configurationname="sofia.conf"description="Sofia Endpoint Configuration"> ... <profiles> ... <profilename="internal"> ... <aliases> ... <!-- SIP Trunk --> <aliasmatch="[1-9]XXXXXXXXX"> <!-- Dial via provider --> <!--<action application="set" data="__sip_provider=${sip_pro...
PBX -> PBX Configuration -> Extensions 分機號:199 註冊密碼:199pass PS.這裡的步驟與一般分機設置相同 在Asterisk(end) //新增一個 SIP Trunk 註冊於 Asterisk(provider) Trunk Name: ast_provider PEER Details: username=199 type=peer secret=199pass insecure=very host=192.168.1.1 fromuser=199 qualify=...
DIDforSale provides complete support in configuration of SIP Trunk and FreeSWITCH. A brief architecture of the big picture will help you understand what role will FreeSWITCH play in your communication application? FreeSWITCH is also commonly referred to as PBX (Private Branch Exchange). However ...
With the root configuration directory located at/etc/freeswitch/, you must complete the following configurations: Create a new SIP Profile. Create a Dial Plan. Step 1: Creating a SIP Profile Gateway Create a new file named “zentrunk.xml” at /etc/freeswitch/sip_profiles/external/. ...
Whether it's a "user" or a "trunk" really doesn't matter - it's just a SIP registration. Look in conf/directory/default/1000.xml (if using the example "vanilla" configuration) for a taste of what you need. Let us know if you have any other questions or join us in #freeswitch ...
1> 添加网关配置:如添加[123456]的网关:/usr/local/freeswitch/conf/sip_profiles/external/123456.xml,配置内容如下 <include> <gateway name="123456">
FreeSwitch is a high-performance VoIP/SIP PBX/Switch software package. FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. Installing FreeSwitch Dependencies Things like ODBC and a...
</configuration> acl配置生效 freeswitch@internal> reloadacl reloadxml acl测试 &呼叫测试 1. Test an IP address against one of your ACL's freeswitch@internal> acl 42.96.203.28 domains true freeswitch@internal> 2. 88885000 拨 1010 直接呼叫(注意不是输入sip uri 1010@122.112.86.102) ...
In the Vanilla example configuration there are two default categories for where a SIP Profile can reside, named "internal" and "external", each serving a broad, general purpose.External SIP profile is generally used to communicate with your PSTN gateway or "SIP trunk" service provider, such as...
>> to do it that way, but I'd prefer to have it work with the sip_use_gateway >> scheme you mention. I'm assuming I'm just >> doing >> something wrong with how sip_use_gateway should be specified in the XML >> configuration files. Can you tell what I am ...