Internet Voice – PhoeNIXBird Networks

[NETWORKING] 90’s style ISP with a CISCO 2600

June 1, 2024 Andrei Rachita 0

Some years passed, and I thought I can do better than the setup described in this post: [Linux/Network] PPP/PAP Dial-Up server with VoIP and Raspberry […]

[Linux/Network] PPP/PAP Dial-Up server with VoIP and Raspberry Pi

June 12, 2020 Andrei Rachita 0

OK, so for today’s relaxing session, I though I should setup a PPP/PAP Dial-up server using a Raspberry Pi 3 box. So here’s the hardware: […]

[Linux/Azure/Networking] Host your own open-source video conferencing solution

April 11, 2020 Andrei Rachita 0

In these times when we need to communicate online more than ever, I though I should share how to setup an open-source solution for Video […]

Renew Elastix SSL Certificate

April 13, 2014 Andrei Rachita 0

You want to renew / change your SSL certificate on your Elastix box: [root@localhost ~]# cd / [root@localhost /]# mkdir /etc/httpd/ssl [root@localhost /]# cd /etc/httpd/ssl […]

Install G729 codec and license on Elastix (Asterisk)

November 6, 2013 Andrei Rachita 0

Install G729 codec and license on Elastix 2.x Assuming we are using the 64-bit version of Asterisk 1.8.x 1) First, acquire a G729 codec license […]

How to insert a previous backup on Elastix

June 13, 2013 Andrei Rachita 0

Scenario: You installed a new Elastix 2.4.0 and would like to restore backups from the old installation. Resolution: you access https://serverip/index.php?menu=backup_restore and you can notice that […]

Solving Asterisk 1.8 MOH Issue

August 10, 2012 Andrei Rachita 0

Setup: Asterisk 1.8 + FreeBBX 2.5+ Scenario: you just uploaded some new MOH files, but Asterisk still plays the default ones Solution: – SSH to […]

How to check your VoIP codec in Asterisk

May 26, 2012 Andrei Rachita 0

Scenario: – you want to know what codecs are used when you make a phone call via your Asterisk PBX Resolution: – SSH to your […]

How to delete Asterisk voice mails from CLI

October 22, 2011 Andrei Rachita 0

What if you ask your Asterisk admin to delete all your voice messages on the system ? Here is how the admin will do it: […]

Provisioning a ZyXEL P2002 with Asterisk

August 5, 2011 Andrei Rachita 0

Many people have struggled to make this device work with Asterisk PBX and faced peer registration issues or one-way calls.

Here is a sample configuration that works:

<strong>On Asterisk (version 1.6): (sip_aditional.conf)</strong>

[222]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1111
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=222@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/222
context=from-internal
canreinvite=yes
callgroup=
callerid=device &lt;222&gt;
accountcode=
call-limit=50

On the ZyXEL P2002:
- SIP Account: SIP1 ACTIVE
- SIP Number: 222
- SIP Local Port: 8000
- SIP Server Address: 192.168.1.10
- SIP Server Port: 5060
- REGISTER Server Address: 192.168.1.10
- REGISTER Server Port: 5060
- SIP Service Domain: 192.168.1.10
- Authentication ID: 222
- Authentication Password: 1111
- Sending Called ID: CHECKED
- Incoming Calls Apply to: PHONE 1

ADVANCED SETTINGS:
- URL Type: SIP
Leave the rest on default values
- Caller Ringing: ENABLED with Default Tone

If everything is OK, at the end, you should have the following statuses:

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Here is a sample configuration that works:

On Asterisk (version 1.6): (sip_aditional.conf)

[222]

deny=0.0.0.0/0.0.0.0

type=friend

secret=1111

qualify=yes

port=5060

pickupgroup=

permit=0.0.0.0/0.0.0.0

nat=yes

mailbox=222@device

host=dynamic

dtmfmode=rfc2833

dial=SIP/222

context=from-internal

canreinvite=yes

callgroup=

callerid=device <222>

accountcode=

call-limit=50

On the ZyXEL P2002:

- SIP Account: SIP1 ACTIVE

- SIP Number: 222

- SIP Local Port: 8000

- SIP Server Address: 192.168.1.10

- SIP Server Port: 5060

- REGISTER Server Address: 192.168.1.10

- REGISTER Server Port: 5060

- SIP Service Domain: 192.168.1.10

- Authentication ID: 222

- Authentication Password: 1111

- Sending Called ID: CHECKED

- Incoming Calls Apply to: PHONE 1

ADVANCED SETTINGS:

- URL Type: SIP

Leave the rest on default values

- Caller Ringing: ENABLED with Default Tone

If everything is OK, at the end, you should have the following statuses:

<strong>On the ZyXEL P2002: (Maintenance/Status)</strong>
SIP1 Registration Status: On Register
Used Port: 8000/UDP

<strong>On Asterisk:</strong>

[PBX.local ~]# asterisk -vvvrrrrr
Asterisk 1.6.0.26-FONCORE-r78, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer &lt;markster@digium.com&gt;
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on pbnet (pid = 2693)
Verbosity is at least 6
pbnet*CLI&gt; !clear
pbnet*CLI&gt; sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
SIP TRUNK/001001001        xx.xx.xxx.xx                5060     OK (6 ms)
221/221                    192.168.1.90     D   N   A  5060     OK (12 ms)
222/222                    192.168.1.20     D   N   A  8000     OK (12 ms)
223/223                    192.168.1.21     D   N   A  5060     OK (14 ms)
224/224                    (Unspecified)    D   N   A  5060     (UNKNOWN)
5 sip peers [Monitored: 4 online, 1 offline Unmonitored: 0 online, 0 offline]
PBX*CLI&gt;

Note: extension 224 is offline

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On the ZyXEL P2002: (Maintenance/Status)

SIP1 Registration Status: On Register

Used Port: 8000/UDP

On Asterisk:

[PBX.local ~]# asterisk -vvvrrrrr

Created by Mark Spencer <markster@digium.com>

Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.

This is free software, with components licensed under the GNU General Public

License version 2 and other licenses; you are welcome to redistribute it under

certain conditions. Type 'core show license' for details.

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

== Parsing '/etc/asterisk/asterisk.conf': == Found

== Parsing '/etc/asterisk/extconfig.conf': == Found

Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on pbnet (pid = 2693)

Verbosity is at least 6

pbnet*CLI> !clear

pbnet*CLI> sip show peers

Name/username Host Dyn Nat ACL Port Status

SIP TRUNK/001001001 xx.xx.xxx.xx 5060 OK (6 ms)

221/221 192.168.1.90 D N A 5060 OK (12 ms)

222/222 192.168.1.20 D N A 8000 OK (12 ms)

223/223 192.168.1.21 D N A 5060 OK (14 ms)

224/224 (Unspecified) D N A 5060 (UNKNOWN)

5 sip peers [Monitored: 4 online, 1 offline Unmonitored: 0 online, 0 offline]

PBX*CLI>

Note: extension 224 is offline

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