Sofia SIP Stack - FreeSWITCH - Confluence
SIP.js Update: Video Conferencing & Secure Calling Added SIP.js allows you to build web apps to allow these SIP user agents to make voice and video calls over the Internet or to the public switched telephone network (PSTN). FreeSWITCH is a critical component of OnSIP's network architecture. Although SIP.js supports other platforms, such as Asterisk, many of our SIP servers run on FreeSWITCH. freeswitch-users - 407 Proxy Authentication Required 407 Proxy Authentication Required. I'm trying to pass a sip call from ProxyA to my Freeswitch but my FS always return a "407 Proxy Authentication Required" to ProxyA as follow Source Voxbone | Using Voxbone with FreeSWITCH
407 Proxy Authentication Required. I'm trying to pass a sip call from ProxyA to my Freeswitch but my FS always return a "407 Proxy Authentication Required" to ProxyA as follow Source
Neither kamailio or freeswitch are an SBC. Freeswitch and Asterisk are b2bua and ser/kamailio/opensips is a proxy. I'd use Kamailio in your case (prefer over opensips, but that's a long story) and either use rtpproxy to proxy media or, since you're not, just use as a proxy with either LCR … SIP.js Update: Video Conferencing & Secure Calling Added
Setting CID Method. In the channel: Channel_Variables#Caller_ID_Related, specifically: sip_cid_type In the gateway: Sofia Gateway Authentication Params- you can set
sip - Freeswitch wrong caller id number after bridge