Install FreePBX

Add port UDP 5060 NAT entry in firewall

Set up new user (extension)
Applications >> Extensions >> Add new Extension JPSIP (newer protocol) >> 800 >> Secret=P@ssword >> Password For New User = P@ssword >> Voicemail >> Enabled, password=12345 >> Email address=kdoan@westernmutual.com >> require from same extension=no, email attachment=yes, delete voicemail=yes >> submit >> apply config

show on server console
asterisk -rx "pjsip show endpoints"

Yate Client: Server=12.46.12.139, Authentication username=800

If not already available, install g729 (g.729E MOS score 4.2, g.729a MOS score 3.7)
check: asterisk -rx "core show codecs"
check processor architecture: "less /proc/cpuinfo"
check asterisk version: asterisk -r
get codec here: http://asterisk.hosting.lv/#bin
cd /usr/lib64/asterisk/modules
wget http://asterisk.hosting.lv/bin/codec_g729-ast130-gcc4-glibc2.2-x86_64-core2-sse4.so
mv codec_g729-ast130-gcc4-glibc2.2-x86_64-core2-sse4.so format_g729e.so
/etc/init.d/asterisk restart
check result: asterisk -rx "core show codecs"

Optional:
Disable root login
useradd kim
passwd kim
vim /etc/ssh/sshd_config
------------
#PermitRootLogin no
AllowUsers kim
------------
visudo
kim ALL=(ALL) ALL
------------
Other essentials
yum install gcc java fail2ban

Load PbxInAFlash.ova into VMWare
Install.... then access WEB interface...
SIP: TCP 5060-5082
Calling: UDP 10001-20000
Google Voice: TCP 5222
Log onto FreePBX >> Edit NAT settings: external IP=FQDN, Local networks=192.168.x.0/24
Codecs: ulaw 8khz (max with Google), g729 (popular), g722 (HD)
Applications >> Extension >> generic sip device >> Add SIP Extension
example:
User Extension = 201
Display name = Kim
CID Num Alias = 7142027660 (Google Voice number)
SIP Alias = Kim
Outbound CID = 7142027660
Secret = {change it to something you remember}
submit >> Apply Config >> go back
qualify = no (this phone doesn't have to constantly ping server for keepalive)
Add Google voice account
Connectivity >> Google Voice [motif] >> enter google info, add trunk, add outbound routes, send unanswered calls to voicemail
Connectivity >> outbound routes >> Dial patterns: add +1NXXNXXXXXX as third entry, callerID=Google Number (7142027660) >> submit changes >> apply
Connectivity >> inbound routes >> DID number = 7142027660 >> Enable CID Superfecta, check >> set destination, extensions, <201> Kim >> submit >> apply
ssh into Linux OS >> amportal restart
goto https://code.google.com/p/csipsimple >> click on nightly builts: http://nightlies.csipsimple.com/trunk >> get version 2174 (last version that has g729 codec) >> access the Android phone (vmware image) >> install the apk >> start VSipSimple >> click and hold account >> go to Advanced >> set server, username=201, proxy, enable QoS, enable SIP, Enable Stun=enable (for wireless carrier data interupts), clock rate=8Khz (voice quality), codecs:fast=g722,slow=g729 >> Filters: User mobile, set 911 to use it >> test call *43 (echo test)
goto FreePBX server site >> FreePBX System Status

How to install g729 codec:
Download G729 codec from http://asterisk.hosting.lv
transfer the codec to server using WinSCP
destination: /usr/lib/asterisk/modules
ssh: amportal restart
Use Softphone to test
How to apply updates:
Admin >> Backup & Restore >> enter name as date >> drag and drop "full backup" >> choose storage server >> submit
Save it on desktop:
admin >> Backup & restore >> restore >> choose files >> browse=choose Desktop >> Go
Admin >> FreePBX Support? >> check online >> check next to show only upgradable >> select upgrades >> apply config

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