Sip-Tools - Automated calls and answering machine

Sip-Tools - 라즈베리 파이를 SIP 자동응답기로 만드는 방법 (pjsip client 사용)

major git https://github.com/fabianhu/SIP-Pi

updated 2017 06 10 by OLSSOO FACTORY

norman.southcastle@gmail.com

준비

SIP 계정 (내선번호도 가능)
라즈베리 파이 2/3
사운드 파일 (wav형식)
1 ~ 9번 입력시 번호별 동작 계획(DTMF)

설치순서

PJSIP
RAS-SIP-Pi

=================================================

SIP 설정샘플

cat sipserv.cfg 
sr=OLSSOO
sd=101.250.108.143:55520
su=3651
sp=3651ok!

네이버 음성합성 Open API PHP/Python 파일 추가 (mp3을 wav파일로 변환 후 사용)

naver-tts-v1.php naver-tts-v1.py

Sip-Tools - Automated calls and answering machine

sipcall - Automated calls over SIP/VOIP with TTS
sipserv - Answering machine for SIP/VOIP with TTS

Dependencies:

PJSUA API (http://www.pjsip.org)
eSpeak (http://espeak.sourceforge.net)

major changes 2017 by Fabian Huslik, github.com/fabianhu

For more informations please visit http://binerry.de/post/29180946733/raspberry-pi-caller-and-answering-machine.

Installation on Raspberry Pi 2/3 with Raspian

Build and install PjSIP as explained below
install eSpeak sudo apt-get install espeak espeak-data
Copy Project folder to Raspberry Pi and hitmake in this folder
configure sipserv.cfg to your needs (see example configuration)
test drive using./sipserv --config-file sipserv.cfg
this is not(yet) a "real" service, so include ./sipserv-ctrl.sh start command into your favourite autostart.
stop the SIP service using sipserv-ctrl.sh stop
install lame sudo apt-get install lame for the MP3 compression of recordings (mail.sh)

sipserv

Pickup a call, have a welcome message played or read. Do some actions by pressing (DTMF) keys on your phone. This service uses a generic approach. All actions are configurable via config file. One special usage is the special ability to record the caller while playing the intro. Please contact your lawyer, if this is legal in your country. With the sample configuration you can have a blacklist and only the special (=blacklisted) calls answered.

##Usage:
sipserv [options]

##Commandline:
###Mandatory options:

--config-file=string Set config file

###Optional options:

-s=int Silent mode (hide info messages) (0/1)

##Config file:
###Mandatory options:

sr=string Set sip realm.
sd=string Set sip provider domain.
su=string Set sip username.
sp=string Set sip password.
ln=string Language identifier for espeak TTS (e.g. en = English or de = German).
tts=string String to be read as a intro message

###and at least one dtmf configuration (X = dtmf-key index):

dtmf.X.active=int Set dtmf-setting active (0/1).
dtmf.X.description=string Set description.
dtmf.X.tts-intro=string Set tts intro.
dtmf.X.tts-answer=string Set tts answer.
dtmf.X.cmd=string Set shell command.

###Optional options:

rc=int Record call (0=no/1=yes)
af=string announcement wav file to play; tts will not be read, if this parameter is given. File format is Microsoft WAV (signed 16 bit) Mono, 22 kHz;
cmd=string command to check if the call should be taken; the wildcard # will be replaced with the calling phone number; should return a "1" as first char, if you want to take the call.
am=string aftermath: command to be executed after call ends. Will be called with two parameters: $1 = Phone number $2 = recorded file name

##a sample configuration can be found in sipserv-sample.cfg

##sipserv can be controlled with

./sipserv-ctrl.sh start and 
./sipserv-ctrl.sh stop

Build PjSIP

build directly on Raspberry Pi:

https://www.raspberrypi.org/forums/viewtopic.php?f=28&t=178384

Install dependencies

sudo apt-get install libasound2-dev libssl-dev libv4l-dev libsdl2-dev libsdl2-gfx-dev libsdl2-image-dev libsdl2-mixer-dev libsdl2-net-dev libsdl2-ttf-dev libx264-dev libavformat-dev libavcodec-dev libavdevice-dev libavfilter-dev libavresample-dev libavutil-dev libavcodec-extra-56 libopus-dev libopencore-amrwb-dev libopencore-amrnb-dev libvo-amrwbenc-dev subversion gobjc++ mpg321 sox libsox-fmt-mp3 php5 php5-curl

Create a directory for the source code

mkdir -p ~/usr/src/olssoo && cd ~/usr/src/olssoo

Build and install OpenH264

wget https://github.com/cisco/openh264/archive/v1.6.0.tar.gz
tar -xf v1.6.0.tar.gz
cd openh264-1.6.0
make
sudo make install
cd ..

Download and extract PJSIP source

svn checkout http://svn.pjsip.org/repos/pjproject/trunk
cd trunk
./configure 
make dep 
make
sudo make install

Enable video support

pi@raspberrypi2b:~/software/source/pjproject-2.6 $ nano pjlib/include/pj/config_site.h
pi@raspberrypi2b:~/software/source/pjproject-2.6 $ cat pjlib/include/pj/config_site.h
#define PJMEDIA_HAS_VIDEO 1

Set compiler options

pi@raspberrypi2b:~/software/source/pjproject-2.6 $ nano user.mak
pi@raspberrypi2b:~/software/source/pjproject-2.6 $ cat user.mak
# You can create user.mak file in PJ root directory to specify
# additional flags to compiler and linker. For example:
export CFLAGS += -march=armv7-a -mfpu=neon-vfpv4 -ffast-math -mfloat-abi=hard
export LDFLAGS +=

Modify third_party/build/os-auto.mak.in (ing)

pi@raspberrypi2b:~/software/source/pjproject-2.6 $ nano third_party/build/os-auto.mak.in
pi@raspberrypi2b:~/software/source/pjproject-2.6 $ cat third_party/build/os-auto.mak.in

ifneq (@ac_no_gsm_codec@,1)
ifeq (@ac_external_gsm@,1)
# External
else
DIRS += gsm
endif
endif

ifneq (@ac_no_ilbc_codec@,1)
DIRS += ilbc
endif

ifneq (@ac_no_speex_codec@,1)
ifeq (@ac_external_speex@,1)
# External speex
else
DIRS += speex
endif
endif

ifneq (@ac_no_g7221_codec@,1)
DIRS += g7221
endif

ifneq ($(findstring pa,@ac_pjmedia_snd@),)
ifeq (@ac_external_pa@,1)
# External PA
else
#DIRS += portaudio
endif
endif

ifeq (@ac_external_srtp@,1)
# External SRTP
else
DIRS += srtp

ifeq (@ac_ssl_has_aes_gcm@,0)
CIPHERS_SRC = crypto/cipher/aes.o crypto/cipher/aes_icm.o       \
              crypto/cipher/aes_cbc.o
HASHES_SRC  = crypto/hash/sha1.o crypto/hash/hmac.o       \
         # crypto/hash/tmmhv2.o
RNG_SRC     = crypto/rng/rand_source.o crypto/rng/prng.o    \
         crypto/rng/ctr_prng.o
else
CIPHERS_SRC = crypto/cipher/aes_icm_ossl.o crypto/cipher/aes_gcm_ossl.o
HASHES_SRC  = crypto/hash/hmac_ossl.o
RNG_SRC     = crypto/rng/rand_source_ossl.o
SRTP_OTHER_CFLAGS = -DOPENSSL
endif


endif

ifeq (@ac_pjmedia_resample@,libresample)
DIRS += resample
endif

ifneq (@ac_no_yuv@,1)
ifeq (@ac_external_yuv@,1)
# External yuv
else
DIRS += yuv
endif
endif

ifneq (@ac_no_webrtc@,1)
ifeq (@ac_external_webrtc@,1)
# External webrtc
else
DIRS += webrtc
WEBRTC_OTHER_CFLAGS = -fexceptions -DWEBRTC_POSIX=1 @ac_webrtc_cflags@
#ifneq ($(findstring sse2,@ac_webrtc_instset@),)
#    WEBRTC_SRC = \
#             modules/audio_processing/aec/aec_core_sse2.o       \
#         modules/audio_processing/aec/aec_rdft_sse2.o            \
#         modules/audio_processing/aecm/aecm_core_c.o            \
#         modules/audio_processing/ns/nsx_core_c.o                    \
#         system_wrappers/source/cpu_features.o
#else ifneq ($(findstring neon,@ac_webrtc_instset@),)
WEBRTC_SRC = \
    modules/audio_processing/aec/aec_core_neon.o               \
    modules/audio_processing/aec/aec_rdft_neon.o               \
    modules/audio_processing/aecm/aecm_core_c.o                \
    modules/audio_processing/aecm/aecm_core_neon.o             \
    modules/audio_processing/ns/nsx_core_c.o                   \
    modules/audio_processing/ns/nsx_core_neon.o                \
    common_audio/signal_processing/cross_correlation_neon.o    \
    common_audio/signal_processing/downsample_fast_neon.o      \
    common_audio/signal_processing/min_max_operations_neon.o
WEBRTC_OTHER_CFLAGS += -DWEBRTC_HAS_NEON
#else ifneq ($(findstring mips,@ac_webrtc_instset@),)
#    WEBRTC_SRC = \
#              modules/audio_processing/aec/aec_core_mips.o               \
#         modules/audio_processing/aec/aec_rdft_mips.o               \
#         modules/audio_processing/aecm/aecm_core_mips.o             \
#         modules/audio_processing/ns/nsx_core_mips.o                \
#         common_audio/signal_processing/cross_correlation_mips.o    \
#         common_audio/signal_processing/downsample_fast_mips.o      \
#         common_audio/signal_processing/min_max_operations_mips.o
#
#    WEBRTC_OTHER_CFLAGS += -DMIPS_FPU_LE
#else # Generic fixed point
#    WEBRTC_SRC = \
#         modules/audio_processing/aecm/aecm_core_c.o                \
#         modules/audio_processing/ns/nsx_core_c.o                   \
#         common_audio/signal_processing/complex_fft.o
#endif
endif
endif

You will have plenty of time to brew some coffe during make. Enjoy while waiting.

Cross build of PjSIP for Raspberry:

export CC=/opt/raspi_tools/tools/arm-bcm2708/gcc-linaro-arm-linux-gnueabihf-raspbian-x64/bin/arm-linux-gnueabihf-gcc
export LD=/opt/raspi_tools/tools/arm-bcm2708/gcc-linaro-arm-linux-gnueabihf-raspbian-x64/bin/arm-linux-gnueabihf-gcc
export CROSS_COMPILE=/opt/raspi_tools/tools/arm-bcm2708/gcc-linaro-arm-linux-gnueabihf-raspbian-x64/bin/arm-linux-gnueabihf-
#export AR+=" -rcs"

export LDFLAGS="-L/opt/raspi_tools/tools/arm-bcm2708/gcc-linaro-arm-linux-gnueabihf-raspbian-x64/lib/gcc/arm-linux-gnueabihf/4.8.3 -L/opt/raspi_tools/tools/arm-bcm2708/gcc-linaro-arm-linux-gnueabihf-raspbian-x64/arm-linux-gnueabihf/lib -ldl -lc"

./aconfigure --host=arm-elf-linux --prefix=$(pwd)/tmp_build --disable-video 

make dep

make

sipcall

Make outgoing calls with your Pi.

##Usage:

sipcall [options]

##Mandatory options:

sr=string Set sip realm.
-sd=string Set sip provider domain.
-su=string Set sip username.
-sp=string Set sip password.
-pn=string Set target phone number to call
-tts=string Text to speak

##Optional options:

-ttsf=string TTS speech file name
-rcf=string Record call file name
-mr=int Repeat message x-times
-s=int Silent mode (hide info messages) (0/1)

see also source of sipcall-sample.sh

License

This tools are free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version.

This tools are distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details.

Name		Name	Last commit message	Last commit date
Latest commit History 75 Commits
.idea		.idea
.project		.project
Makefile		Makefile
README.md		README.md
ansage.wav		ansage.wav
cp_mk.sh		cp_mk.sh
mail.py		mail.py
mail.sh		mail.sh
naver-tts-v1.php		naver-tts-v1.php
naver-tts-v1.py		naver-tts-v1.py
numbers.txt		numbers.txt
numcheck.py		numcheck.py
olssoo_connect.wav		olssoo_connect.wav
sipcall		sipcall
sipcall-sample.sh		sipcall-sample.sh
sipcall.c		sipcall.c
sipserv		sipserv
sipserv-ctrl.sh		sipserv-ctrl.sh
sipserv-sample.cfg		sipserv-sample.cfg
sipserv.c		sipserv.c
sipserv.cfg		sipserv.cfg

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Sip-Tools - Automated calls and answering machine

Installation on Raspberry Pi 2/3 with Raspian

sipserv

Build PjSIP

build directly on Raspberry Pi:

최신버젼의 pjsip 설치

https://www.raspberrypi.org/forums/viewtopic.php?f=28&t=178384

Cross build of PjSIP for Raspberry:

sipcall

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Sip-Tools - Automated calls and answering machine

Installation on Raspberry Pi 2/3 with Raspian

sipserv

Build PjSIP

build directly on Raspberry Pi:

최신버젼의 pjsip 설치

https://www.raspberrypi.org/forums/viewtopic.php?f=28&t=178384

Cross build of PjSIP for Raspberry:

sipcall

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