[Nexus 4 General] acegolfer:
Tips/Tricks

Tips/Tricks

1. Use ring group to ring all the extensions, if you have multiple sip phones.

2. Add more GV numbers

My family have 3 GV numbers + 6 phones. I added all 3 gtalk trunks to PIAF. OTOH, pbxes doesn't allow you to add more than 1 gtalk trunk. Since all 6 phones are in the same PBX, I can call my wife's phone by her extension number (or her ring group) instead of her GV number. If you do this, you need to modify outbound routes such that certain extensions use specific GV trunk. Otherwise, all the phones will use the first GV trunk to make outgoing calls and display the same CID.

3. Call forward to external phones.

In the ring group, you can have PIAF call regular phone number such as your cell phone. Just list the phone number followed by #. One problem is gtalk trunk doesn't allow CID spoofing so the external phone will show your GV# as CID.

4. Delayed call forwarding

Create 2 ring groups and put the number for delay in the 2nd group. Set 1st ring group's destination if no answer as the 2nd group and specify the ring time. I do this for my office phone because it goes to voicemail in 20 seconds (before GV voicemail answers). Now my office phone starts ringing after 7 seconds so it will never answer before GV voicemail does. You can also use this strategy with your cell phone such that your cell number rings after you have a chance to answer internet calls.

5. Secure your PIAF

If you followed pbxinaflash.com guide, then you opened port 80 and 22 to the world. Hackers can brute force using username = root. You should limit who can access these ports by IP address. Additionally, you can disable root SSH login and create a sudoer.

http://www.howtogeek.com/howto/linux/security-tip-disable-root-ssh-login-on-linux/

http://wiki.centos.org/TipsAndTricks/BecomingRoot

6. Add SILK codec

If you are on PIAF-green (asterisk 11), then you can add SILK codec. This is the codec that Skype uses so you may get the same excellent quality as Skype. From linux CLI,

wget http://downloads.digium.com/pub/telephony/codec_silk/asterisk-11.0/x86-64/codec_silk-11.0_1.0.0-core2_64.tar.gz
tar -zxvf codec_silk-11.0_1.0.0-core2_64.tar.gz
cd codec_silk-11.0_1.0.0-core2_64
cp codec_silk.so /usr/lib/asterisk/modules/codec_silk.so

Restart asterisk and from asterisk CLI, core show codecs and confirm that 4 silk codecs (silk8, silk12, silk16, silk24) are added. If not, read http://xdaforums.com/showpost.php?p=38398737&postcount=71.

To use SILK codec, From extension, set

disallow=all

allow=silk12

7. Eliminate echo

ryuker has posted a method to eliminate the echo.

http://xdaforums.com/showpost.php?p=42740388&postcount=306

Troubleshooting

There are many many reasons why GV calling fails. This means you will probably not get any help, unless you provide more information such as asterisk log. If you want to troubleshoot by yourself, try these first.

1. csip / piaf registration: Try a different sip client or use UDP connection.

2. GV: See whether you can make GV calls using GrooveIP. GV calling may not work for new accounts until you initiate a call from gmail.

3. Restart asterisk. If you add a GV trunk, it must be restarted to take into effect.

2. piaf / GV link: Add simonics GVGW as a sip trunk. https://simonics.com/gvgw/