[asterisk-commits] rmudgett: branch rmudgett/srtp r1740 - in /asterisk/team/rmudgett/srtp/tests/...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 12 15:58:25 CDT 2011
Author: rmudgett
Date: Tue Jul 12 15:58:22 2011
New Revision: 1740
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=1740
Log:
Address reviewboard comments.
* Removed unneeded tcpenable and tcpbindaddr in sip.conf files.
* Made authenticate all calls.
* Made ast2 use 127.0.0.2 IP address instead of a different port.
* Made secure_bridge_media test use "channel originate" instead of
"console dial" CLI command for better portability on test machines.
Modified:
asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/run-test
asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf
asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf Tue Jul 12 15:58:22 2011
@@ -4,8 +4,6 @@
udpbindaddr=127.0.0.1:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
;encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=notguessable ; The password they use to contact us
+remotesecret=notguessable ; The password we use to authenticate to them
+secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,7 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf Tue Jul 12 15:58:22 2011
@@ -2,10 +2,8 @@
context = siptest2
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
-udpbindaddr=127.0.0.1:5061 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+udpbindaddr=127.0.0.2:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
;encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+remotesecret=youwillneverguessit ; The password we use to authenticate to them
+secret=notguessable ; The password they use to contact us
+;remotesecret=notguessable ; The password we use to authenticate to them
+;secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,8 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
-
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf Tue Jul 12 15:58:22 2011
@@ -4,8 +4,6 @@
udpbindaddr=127.0.0.1:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=notguessable ; The password they use to contact us
+remotesecret=notguessable ; The password we use to authenticate to them
+secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,7 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf Tue Jul 12 15:58:22 2011
@@ -2,10 +2,8 @@
context = siptest2
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
-udpbindaddr=127.0.0.1:5061 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+udpbindaddr=127.0.0.2:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
;encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+remotesecret=youwillneverguessit ; The password we use to authenticate to them
+secret=notguessable ; The password they use to contact us
+;remotesecret=notguessable ; The password we use to authenticate to them
+;secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,8 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
-
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf Tue Jul 12 15:58:22 2011
@@ -4,7 +4,8 @@
[siptest1]
-exten => 1000,1,Set(CHANNEL(secure_bridge_media)=1)
+exten => 1000,1,NoOp(CHANNEL(secure_bridge_media)=${CHANNEL(secure_bridge_media)})
+exten => 1000,n,Set(CHANNEL(secure_bridge_media)=1)
exten => 1000,n,Set(TEST_RESULT=${CHANNEL} secure_media=${CHANNEL(secure_media)})
;No need to report to the test script since this is the wrong channel anyway.
;exten => 1000,n,AGI(agi://127.0.0.1:4573)
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf Tue Jul 12 15:58:22 2011
@@ -4,8 +4,6 @@
udpbindaddr=127.0.0.1:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
;encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=notguessable ; The password they use to contact us
+remotesecret=notguessable ; The password we use to authenticate to them
+secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,7 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf Tue Jul 12 15:58:22 2011
@@ -2,10 +2,8 @@
context = siptest2
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
-udpbindaddr=127.0.0.1:5061 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+udpbindaddr=127.0.0.2:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+remotesecret=youwillneverguessit ; The password we use to authenticate to them
+secret=notguessable ; The password they use to contact us
+;remotesecret=notguessable ; The password we use to authenticate to them
+;secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,8 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
-
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/run-test?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/run-test (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/secure_bridge_media/run-test Tue Jul 12 15:58:22 2011
@@ -39,13 +39,14 @@
TestCase.run(self)
print "Initiating test call"
- self.ast[0].cli_exec("console dial 1000 at siptest1")
+ self.ast[0].cli_exec(
+ "originate Local/1000 at siptest1 application playback tt-monkeys")
# This is called by each Asterisk instance if the call gets connected.
def fastagi_connect(self, agi):
def get_test_result(val):
print "Connection result '%s'" % val
- if val.split(" ")[0] == "Console/dsp":
+ if val.split("-")[0] == "Local/1000 at siptest1":
# Outgoing call on Ast1
self.connected_chan1 = True
if val.split("=")[1] == "1":
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf Tue Jul 12 15:58:22 2011
@@ -4,8 +4,6 @@
udpbindaddr=127.0.0.1:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=notguessable ; The password they use to contact us
+remotesecret=notguessable ; The password we use to authenticate to them
+secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,7 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
+port=5060
Modified: asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf?view=diff&rev=1740&r1=1739&r2=1740
==============================================================================
--- asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf (original)
+++ asterisk/team/rmudgett/srtp/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf Tue Jul 12 15:58:22 2011
@@ -2,10 +2,8 @@
context = siptest2
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
-udpbindaddr=127.0.0.1:5061 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+udpbindaddr=127.0.0.2:5060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
-tcpenable=no ; Enable server for incoming TCP connections (default is no)
-tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
sipdebug=yes
@@ -17,7 +15,7 @@
sendrpid=yes
trustrpid=yes
canreinvite=yes
-insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=invite ; Do not require authentication of incoming INVITEs
nat=no
directmedia=no
@@ -25,6 +23,12 @@
encryption=yes ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+
+; For SIP signaling authentication.
+remotesecret=youwillneverguessit ; The password we use to authenticate to them
+secret=notguessable ; The password they use to contact us
+;remotesecret=notguessable ; The password we use to authenticate to them
+;secret=youwillneverguessit ; The password they use to contact us
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -36,6 +40,7 @@
[1000](sip-trunk,my-codecs)
callerid="Ast1Name" <1000>
+username=2000
host=127.0.0.1 ; we have a static but private IP address
; No registration allowed
@@ -43,8 +48,8 @@
[2000](sip-trunk,my-codecs)
callerid="Ast2Name" <2000>
+username=1000
-host=127.0.0.1 ; we have a static but private IP address
+host=127.0.0.2 ; we have a static but private IP address
; No registration allowed
-port=5061
-
+port=5060
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