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Trace:
wiki:voip:asterisk:sip_trunk
====== SIP Trunk ====== ===== Server A ===== <file - sip.conf> [serverA] type = peer host = 192.168.1.101 username = serverB secret = apples context = incoming disallow = all allow = alaw </file> <file - extensions.conf> exten => _5XXX,1,Dial(SIP/${EXTEN}@serverA) </file> ===== Server B ===== <file - sip.conf> [serverB] type = peer host = 192.168.1.102 username = serverA secret = apples context = incoming disallow = all allow = alaw </file> <file - extensions.conf> exten => _6XXX,1,Dial(SIP/${EXTEN}@serverB) </file> ===== Trunk s registrací ===== <file - sip.conf> [general] ... register => username:password@your.provider.tld ... [myprovider] type = peer host = your.provider.tld username = username secret = password ; Most providers won't authenticate when they send calls to you, ; so you need this line to just accept their calls. insecure = invite dtmfmode = rfc2833 disallow = all allow = alaw </file> <file - extensions.conf> exten => _XXXXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) </file> ===== Šifrování hovorů (SIP TLS, SRTP) ===== FIXME str.150
wiki/voip/asterisk/sip_trunk.txt
· Last modified: 2014/12/26 18:31 (external edit)
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