Built motion from commit de2cb856.|2.0.33

[motion2.git] / public / app / main / apps / tools / i18n / en.json

diff --git a/public/app/main/apps/tools/i18n/en.json b/public/app/main/apps/tools/i18n/en.json
index 67ff697..bd7165f 100644 (file)
--- a/public/app/main/apps/tools/i18n/en.json
+++ b/public/app/main/apps/tools/i18n/en.json
@@ -50,7 +50,7 @@
             "CALL_LIMIT": "The limit number of the inbound and outbound calls at a time.",
             "DIRECTMEDIA": "Asterisk by default tries to redirect the RTP media stream to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT). The default setting is YES. If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off.",
             "CALLCOUNTER": "Enable call counters on devices.",
-            "ADD_PHONE_TO_URI": "Define if the provider requires ';user=phone' on URI.",
+            "ADD_PHONE_TO_URI": "Define if the provider requires ';user=phone' on URL.",
             "TRUST_REMOTE_PARTY_ID": "Define if Remote-Party-ID should be trusted.",
             "SEND_REMOTE_PARTY_ID_HEADER": "Define if Remote-Party-ID should be sent (defaults to no).",
             "ENCRYPTION": "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.",
@@ -80,7 +80,7 @@
         "FROMDOMAIN": "From Domain",
         "FROMUSER": "From User",
         "OUTBOUNDPROXY": "Outbound Proxy",
-        "ADD_PHONE_TO_URI": "Add Phone to URI",
+        "ADD_PHONE_TO_URI": "Add Phone to URL",
         "TRUST_REMOTE_PARTY_ID": "Trust Remote Party ID",
         "SEND_REMOTE_PARTY_ID_HEADER": "Send Remote Party ID Header",
         "ENCRYPTION": "Encryption",