commit ef5fc39099d760490f648ac068f72ce8d7b7b2c7
parent de9df97c239128eb35cc77ae8ef8d23753595e9e
Author: Dominik Schmidt <dominik@schm1dt.ch>
Date: Sat, 31 Aug 2019 20:48:50 +0200
Rename the readme to uppercase
Diffstat:
README.md | | | 70 | ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ |
Readme.md | | | 70 | ---------------------------------------------------------------------- |
2 files changed, 70 insertions(+), 70 deletions(-)
diff --git a/README.md b/README.md
@@ -0,0 +1,70 @@
+SiProc
+======
+
+SiProc is a primitive SIP Client used for call automatization by spawning a given process for each call.
+It uses the [pjsip](http://www.pjsip.org/) library as a backend.
+
+Usage
+-----
+
+The main process can be started as follows:
+
+```
+$ export SIPROC_REGISTRAR_URI="sip:myserver.lan"
+$ export SIPROC_ID_URI="Full Name <sip:Account@myserver.lan>"
+$ export SIPROC_USERNAME="Account"
+$ export SIPROC_PASSWORD="secret"
+$ ./siproc /path/to/executable args...
+```
+
+Additionally, one may define any of the following, which default to the values in the readme:
+
+```
+$ export SIPROC_TRANSPORT_PORT="5060"
+```
+
+### Main Control
+
+The siproc executable takes multiple commands on stdin:
+
+* `QUIT`: Shuts down the program
+* `CALL sip:contact@remote`: Call the given sip URI
+
+### Calls
+
+Whenever a call is made or received, `/path/to/executable` is executed, and is given the following environment variables
+
+* SIPROC_REMOTE_URI: The address of the caller
+* SIPROC_REMOTE_ID: Some pjsip interna, look up their documentation
+* SIPROC_REMOTE_CONTACT: Ditto
+* SIPROC_OUTGOING: Whether a call is being made (SIPROC_OUTGOING="y") or received (SIPROC_OUTGOING="n")
+
+The program can then perform actions via STDIO.
+Each command is separated by a newline ('\n').
+The following commands are supported:
+
+* `RINGING`: Let's the remote end know that we are ringing.
+* `ANSWER`: Answer an incoming call
+* `HANGUP`: Hang up a call
+* `TRANSFER sip:contact@remote`: Redirect a call
+* `DTMF string`: Dial the DTMF tones of each character in `string`
+* `PLAY </path/to/file.wav>`: Play a file. Multiple files can be played at the same time.
+* `STOP`: Stop the file that has been added last.
+* `MESSAGE <msg>`: Send a SIP message to the remote side
+* `RECORD </path/to/file.wav>`: Record the conversation to the given file. If no file is given, the current recording is stopped.
+
+The following lines can be received via stdin:
+
+* `DTMF c`: When the client receives the DTMF character `c`
+* `CONNECTED`: When the remote end has picked up the phone
+* `STOPPED /path/to/file.wav`: Playback of the given file has been terminated.
+* `MESSAGE <msg>`: The remote end has sent a message.
+
+When the remote end hangs up, the process is killed via SIGTERM, so you might want to catch that signal for cleaning up.
+
+To-Do
+-----
+
+* [ ] Expose more of pjsips features to the process!
+* [ ] Allow connecting to alsa devices
+
diff --git a/Readme.md b/Readme.md
@@ -1,70 +0,0 @@
-SiProc
-======
-
-SiProc is a primitive SIP Client used for call automatization by spawning a given process for each call.
-It uses the [pjsip](http://www.pjsip.org/) library as a backend.
-
-Usage
------
-
-The main process can be started as follows:
-
-```
-$ export SIPROC_REGISTRAR_URI="sip:myserver.lan"
-$ export SIPROC_ID_URI="Full Name <sip:Account@myserver.lan>"
-$ export SIPROC_USERNAME="Account"
-$ export SIPROC_PASSWORD="secret"
-$ ./siproc /path/to/executable args...
-```
-
-Additionally, one may define any of the following, which default to the values in the readme:
-
-```
-$ export SIPROC_TRANSPORT_PORT="5060"
-```
-
-### Main Control
-
-The siproc executable takes multiple commands on stdin:
-
-* `QUIT`: Shuts down the program
-* `CALL sip:contact@remote`: Call the given sip URI
-
-### Calls
-
-Whenever a call is made or received, `/path/to/executable` is executed, and is given the following environment variables
-
-* SIPROC_REMOTE_URI: The address of the caller
-* SIPROC_REMOTE_ID: Some pjsip interna, look up their documentation
-* SIPROC_REMOTE_CONTACT: Ditto
-* SIPROC_OUTGOING: Whether a call is being made (SIPROC_OUTGOING="y") or received (SIPROC_OUTGOING="n")
-
-The program can then perform actions via STDIO.
-Each command is separated by a newline ('\n').
-The following commands are supported:
-
-* `RINGING`: Let's the remote end know that we are ringing.
-* `ANSWER`: Answer an incoming call
-* `HANGUP`: Hang up a call
-* `TRANSFER sip:contact@remote`: Redirect a call
-* `DTMF string`: Dial the DTMF tones of each character in `string`
-* `PLAY </path/to/file.wav>`: Play a file. Multiple files can be played at the same time.
-* `STOP`: Stop the file that has been added last.
-* `MESSAGE <msg>`: Send a SIP message to the remote side
-* `RECORD </path/to/file.wav>`: Record the conversation to the given file. If no file is given, the current recording is stopped.
-
-The following lines can be received via stdin:
-
-* `DTMF c`: When the client receives the DTMF character `c`
-* `CONNECTED`: When the remote end has picked up the phone
-* `STOPPED /path/to/file.wav`: Playback of the given file has been terminated.
-* `MESSAGE <msg>`: The remote end has sent a message.
-
-When the remote end hangs up, the process is killed via SIGTERM, so you might want to catch that signal for cleaning up.
-
-To-Do
------
-
-* [ ] Expose more of pjsips features to the process!
-* [ ] Allow connecting to alsa devices
-