To set up with sipml5 I had been through the asterisk offiial site and I do recommand you to visit it.
We need to update several config file which are located on /etc/asterisk. Those filename are listed below
- modules.conf
- extensions.conf
- http.conf
- pjsip.conf
- rtp.conf
I have posted how these file looks below with breif explaination.
-
modules.conf: Since we are using
pjsip, we need to stop loadingsip. As both of them cannot be used simultaneously. You can update manually or use the bash script below:sudo sh -c "echo 'noload => chan_sip.so' >> /etc/asterisk/modules.conf" -
extension.conf:Add these things to the
extension.confat the end of the file. If you have just installed a fresh copy of asterisk you can even override the existing code.I have added two extensions, which are in fact dial plans.
- Where
helloworldjust plays the hello-world music when we call in any number - Whereas the
helloworld2, first plays the hello-world and then calls to another number, it also waits for the dtmf and plays its name based on whether the called number is registered one or not.
[helloworld] exten => _X.,1,NoOp(${EXTEN}) same => n,Playback(hello-world) same => n,Hangup() [helloworld2] exten => _X.,1,NoOp(${EXTEN}) same => n,Playback(hello-world) same => n,Dial(PJSIP/${EXTEN},20) same => n,Read(Digits,,) same => n,Playback(you-entered) same => n,SayNumber(${Digits}) - Where
-
http.conf: Please update the file accordingly, or replace if you want.
[general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem -
pjsip.conf:
199is for web based phone3002and3001for sip clients: (like Linphone for desktop and CSipSimle for mobile)This file need to have:
[transport-wss] type=transport protocol=wss bind=0.0.0.0 [199] type=endpoint aors=199 auth=199 use_avpf=yes media_encryption=dtls dtls_ca_file=/etc/asterisk/keys/ca.crt dtls_cert_file=/etc/asterisk/keys/asterisk.pem dtls_verify=fingerprint dtls_setup=actpass ice_support=yes media_use_received_transport=yes rtcp_mux=yes context=helloworld2 disallow=all allow=ulaw allow=opus [199] type=auth auth_type=userpass username=199 password=199@pass1 [199] type=aor max_contacts=1 remove_existing=yes [transport-udp] type=transport protocol=udp bind=0.0.0.0 [3001] type=endpoint context=helloworld2 disallow=all allow=ulaw auth=3001 aors=3001 [3001] type=auth auth_type=userpass password=3001pass username=3001 [3001] type=aor max_contacts=1 remove_existing=yes [3002] type=endpoint context=helloworld2 disallow=all allow=ulaw auth=3002 aors=3002 [3002] type=auth auth_type=userpass password=3002pass username=3002 [3002] type=aor max_contacts=1 remove_existing=yes -
rtp.conf: Need to have these on rtp.conf.
[general] rtpstart=10000 rtpend=20000 icesupport=true stunaddr=stun.l.google.com:19302
###Create Certificates Call the script as such:
cd /usr/local/src/asterisk-13.17.2/contrib/scripts
sudo ./ast_tls_cert -C pbx.example.com -O "My Super Company" -d /etc/asterisk/keys- The "-C" option is used to define our host - DNS name or our IP address.
- The "-O" option defines our organizational name.
- The "-d" option is the output directory of the keys.
- You'll be asked to enter a pass phrase for /etc/asterisk/keys/ca.key, put in something that you'll remember for later.
- This will create the /etc/asterisk/keys/ca.crt file.
- You'll be asked to enter the pass phrase again, and then the /etc/asterisk/keys/asterisk.key file will be created.
- The /etc/asterisk/keys/asterisk.crt file will be automatically generated.
- You'll be asked to enter the pass phrase a third time, and the /etc/asterisk/keys/asterisk.pem, a combination of the asterisk.key and asterisk.crt files, will be created.
- You can then check your /etc/asterisk/keys directory to verify the new files were created, as such:
ls -w 1 /etc/asterisk/keysAnd you should see:
asterisk.crt
asterisk.csr
asterisk.key
asterisk.pem
ca.cfg
ca.crt
ca.key
tmp.cfgYou can reload the asterisk by:
asterisk -rvvvvvvor simply typing reload on Asterisk's cli.
To verify the web server is running, perform:
netstat -an | grep 8089And you should see:
tcp 0 0 0.0.0.0:8089 0.0.0.0:* LISTEN Next, to ensure these modules are loaded by Asterisk, you can perform:
asterisk -rx "module show like crypto"
asterisk -rx "module show like websocket"
asterisk -rx "module show like opus"You should see something similar to:
# asterisk -rx "module show like crypto"
Module Description Use Count Status Support Level
res_crypto.so Cryptographic Digital Signatures 1 Running core
1 modules loaded
# asterisk -rx "module show like websocket"
Module Description Use Count Status Support Level
res_http_websocket.so HTTP WebSocket Support 3 Running extended
res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support 0 Running core
2 modules loaded
# asterisk -rx "module show like opus"
Module Description Use Count Status Support Level
codec_opus.so OPUS Coder/Decoder 0 Running extended
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
Thank you for publishing this
It works to call 3001 (SIP) from the 199 WebRTC user.
However there is a long pause after placing the call in WebRTC until it gets the HelloWorld message.
Any idea why there is a long pause and what can I do to hurry it up?
Also, I can't place calls from 3001 (SIP) to the 199 WebRTC user, the SIP phone says Unsupported media, I guess SIP negotiation fails?