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--- wiki:voip:asterisk:sip_trunk [2012/11/18 10:56]
root
+++ wiki:voip:asterisk:sip_trunk [2014/12/26 18:31] (current)
@@ Line 1: / Line 1: @@
 ====== SIP Trunk ======
-Server A
+===== Server A =====
-<code>
+<file - sip.conf>
 [serverA]
 type = peer
@@ Line 10: / Line 10: @@
 context = incoming
 disallow = all
-allow = ulaw
+allow = alaw
-</code>
+</file>
 <file - extensions.conf>
 exten => _5XXX,1,Dial(SIP/${EXTEN}@serverA)
 </file>
-Server B
+===== Server B =====
-<code>
+<file - sip.conf>
 [serverB]
 type = peer
@@ Line 25: / Line 25: @@
 context = incoming
 disallow = all
-allow = ulaw
+allow = alaw
-</code>
+</file>
 <file - extensions.conf>
 exten => _6XXX,1,Dial(SIP/${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

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