This is a short video on the setting up of extensions on the Elastix Asterisk based IPPX.
Category: Technical
Categories
Elastix 2.4 Released
Elastix.org have announced the release of 2.4 stable.
Key changes are:
Changes in Elastix Framework:
- The instalation of the Elastix system now its much cleaner.
- The Migration to Privileged Scripts its completed. Now, there its no need to use commands such as /bin/touch, /bin/chmod, etc.
- We improve readability on blackmin theme.
- Fixed readout of FreePBX database password.
- The internal jQuery was updated to 1.8.3 .
- Some minor bug fixes for the Elastix Framework.
- Changes in Elastix Addons :
- Correction for Postgresql repo in ARM architecture.
- Some minor bug fixes for Elastix-Addons.
Changes in Elastix Firstboot :
- Make an update of password in manager.conf more robust in the case it falls out of sync with elastix.conf file
- The Cancel option that used to appear in the dialog_password was removed, because if someone pressed, it no allows to continue configuring passwords. Now only appears the Cancel option after the firstboot if its necesary to change the password already seted.
- Some minor bug fixes for Elastix-Firstboot.
Changes in Elastix Email_Admin :
- Change of files owners for more security in the web path. Creation of new helper scripts (s
- pamconfig,remotesmtp,mailman_conig,relayconfig).
- Was made changes in the module email_account in order to better interaction at moment to create a new email account.
- Some minor bug fixes for Elastix-Email_Admin.
Changes in Elastix Fax :
- NEW MODULE Fax Queue.
- Now errors are displayed when the fax job failed to submit and do not ignore them.
- Remove useless code that could potentially error out the module.
- Implementation of fax job cancelation.
Changes in Elastix PBX :
- Add support and features to following phones: Elastix LXP200, Yealink model SIP-T38G, VP530 model, Alcatel Temporis IP800, Escene 620, Fanvil C62, Damall D3310 and Grandstream model GXV280.
- Modified the way of displaying Reasons for Status in module weak keys.
- In module Control Planel was made changes in function showChannel in order to fix bugs in wich the call made through a sip trunk have not been displayed in control panel.Some minor bug fixes for Elastix-PBX.
Changes in Elastix Security:
- The instalation of this module now its much cleaner.
- Change of files owners for more security int he path web path.
- Some bug fixes for Elastix-Security.
Changes in Elastix System :
- Reimplementation of GUI backup and restore operations on top of backupengine.
- Add options to active o inactive services when reboot system in Process Status Applet.
- Some minor bug fixes for Elastix-System.
- Centos version was updated to 5.9
- Kernel version was updated to 2.6.18-348.1.1
- FreePBX version was updated to 2.8.1-16
- Rhino version was updated to 0.99.6-0.b2
- Asterisk version was updated to 1.8.20
- Dadhi version was updated 2.6.1-4
- Amongst others…
For Product details on Elastix see Here
DOWN LOAD AT http://www.elastix.org/index.php/en/downloads/main-distro.html
Categories
VoIP – Per Call Bandwidth
These protocol header assumptions are used for the calculations:
- 40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes) headers.
- Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over Ethernet).
- 6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header.
- 1 byte for the end-of-frame flag on MP and Frame Relay frames.
- 18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check (CRC).
Note: This table only contains calculations for the default voice payload
| Codec Information | Bandwidth Calculations | ||||||||
|---|---|---|---|---|---|---|---|---|---|
| Codec & Bit Rate (Kbps) | Codec Sample Size (Bytes) | Codec Sample Interval (ms) | Mean Opinion Score (MOS) | Voice Payload Size (Bytes) | Voice Payload Size (ms) | Packets Per Second (PPS) | Bandwidth MP or FRF.12 (Kbps) | Bandwidth w/cRTP MP or FRF.12 (Kbps) | Bandwidth Ethernet (Kbps) |
| G.711 (64 Kbps) | 80 Bytes | 10 ms | 4.1 | 160 Bytes | 20 ms | 50 | 82.8 Kbps | 67.6 Kbps | 87.2 Kbps |
| G.729 (8 Kbps) | 10 Bytes | 10 ms | 3.92 | 20 Bytes | 20 ms | 50 | 26.8 Kbps | 11.6 Kbps | 31.2 Kbps |
| G.723.1 (6.3 Kbps) | 24 Bytes | 30 ms | 3.9 | 24 Bytes | 30 ms | 33.3 | 18.9 Kbps | 8.8 Kbps | 21.9 Kbps |
| G.723.1 (5.3 Kbps) | 20 Bytes | 30 ms | 3.8 | 20 Bytes | 30 ms | 33.3 | 17.9 Kbps | 7.7 Kbps | 20.8 Kbps |
| G.726 (32 Kbps) | 20 Bytes | 5 ms | 3.85 | 80 Bytes | 20 ms | 50 | 50.8 Kbps | 35.6 Kbps | 55.2 Kbps |
| G.726 (24 Kbps) | 15 Bytes | 5 ms | 60 Bytes | 20 ms | 50 | 42.8 Kbps | 27.6 Kbps | 47.2 Kbps | |
| G.728 (16 Kbps) | 10 Bytes | 5 ms | 3.61 | 60 Bytes | 30 ms | 33.3 | 28.5 Kbps | 18.4 Kbps | 31.5 Kbps |
| G722_64k(64 Kbps) | 80 Bytes | 10 ms | 4.13 | 160 Bytes | 20 ms | 50 | 82.8 Kbps | 67.6Kbps | 87.2 Kbps |
| ilbc_mode_20(15.2Kbps) | 38 Bytes | 20 ms | NA | 38 Bytes | 20 ms | 50 | 34.0Kbps | 18.8 Kbps | 38.4Kbps |
| ilbc_mode_30(13.33Kbps) | 50 Bytes | 30 ms | NA | 50 Bytes | 30 ms | 33.3 | 25.867 Kbps | 15.73Kbps | 28.8 Kbps |
| Codec Bit Rate (Kbps) | Based on the codec, this is the number of bits per second that need to be transmitted to deliver a voice call. (codec bit rate = codec sample size / codec sample interval). |
|---|---|
| Codec Sample Size (Bytes) | Based on the codec, this is the number of bytes captured by the Digital Signal Processor (DSP) at each codec sample interval. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval). |
| Codec Sample Interval (ms) | This is the sample interval at which the codec operates. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval). |
| MOS | MOS is a system of grading the voice quality of telephone connections. With MOS, a wide range of listeners judge the quality of a voice sample on a scale of one (bad) to five (excellent). The scores are averaged to provide the MOS for the codec. |
| Voice Payload Size (Bytes) | The voice payload size represents the number of bytes (or bits) that are filled into a packet. The voice payload size must be a multiple of the codec sample size. For example, G.729 packets can use 10, 20, 30, 40, 50, or 60 bytes of voice payload size. |
| Voice Payload Size (ms) | The voice payload size can also be represented in terms of the codec samples. For example, a G.729 voice payload size of 20 ms (two 10 ms codec samples) represents a voice payload of 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ] |
| PPS | PPS represents the number of packets that need to be transmitted every second in order to deliver the codec bit rate. For example, for a G.729 call with voice payload size per packet of 20 bytes (160 bits), 50 packets need to be transmitted every second [50 pps = (8 Kbps) / (160 bits per packet) ] |
Bandwidth Calculation Formulas
These calculations are used:
- Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
- PPS = (codec bit rate) / (voice payload size)
- Bandwidth = total packet size * PPS
We offer an economical solution for end users and resellers to monitor their Asterisk and Linux servers.
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We offer different levels of monitoring from simple uptime and email alerts to system load, disk space and channel usage with email and SMS notification. Web panel and firefox/Chrome plugin available to all levels to view system status.
The service is primarily aimed at Asterisk based IPPBX server but we can monitor other Linux based servers and Mitel systems as well. Our checks on Asterisk servers were customised by us to allow easy and secure deployment as we only require SSH access to make checks and this is secured by server keys.
Service levels
Silver Level £10 setup – £2.50 per month £25.00 per year
- Single Server, 4 services from list below & email alerts.
- Ping test
- SIP/IAX Peer availability
- Asterisk channels
- ISDN availability
- Disk Space
- System Load
- Heartbeat Status
- SIP/IAX2 registration status
- Mitel SNMP Alarm status
Gold Level £10 setup per server – £5.00 per month £50.00 per year
- Upto 2 Servers, 4 services per server, email and SMS alerts by subscription
In addition to the silver list:-
- Asterisk Database status
- Custom checks, (cost for design may be inured)
Additional options.
SMS alerts by arrangement, if using Gradwell Numbers and outbound we can integrate with the SMS API
Extra contact £5 setup
Extra server £10 setup £2.50 per month £25 per year
Extra service £5 setup £0.50 per month £5 per year
Partner options are available, Please contact us for details. Pdf download cymon
New version of firmware released for N300 bases, Upgrade to this if on 072 firmware to fix instability issues
– Problem of instability, which occurred only very sporadically with version 72, and reset of base station after intensive usage solved
– Problem with call transfer of an external party to an external target behind Cisco Manager solved
– de telefoongids (Netherlands): online phonebook search is working again
– Security:
· Password is masked in VOIP Wizard, no longer visible in clear text
· PIN entry delayed if user repeatedly enters wrong PIN
– S68H handset: CLIP presentation is working again
– Blind Call Transfer problem solved with Telavox.se and Firmix.at
– URI dialling: Problem with added international/local area codes fixed
– Problem with consultation call and “Use Area Code Numbers for Calls via VoIP” setting fixed
Yealink has announced the release of the latest Firmware V70 for its award winning IP phone SIP-T2X series.
The key feature of this new Firmware V70 is “M7”, also known as the “unified auto-provision template”. With Firmware V70, the configuration files and the deployment methods of T2X, T3X and VP530 have now been unified.
With the deployment of “M7”, end users now no longer need to maintain different templates of T2X, T3x or VP530. In other words, it lowers the learning curve and increases the business efficiency remarkably.
End users can easily convert their old templates of Yealink IP Phone T2X series and T3X series to “M7” through Yealink Configuration Conversion Tool (CCT). Firmware V70 is now available for download free of charge at www.yealink.com.
Categories
Gigaset Dect test mode
Setting test mode on the Gigaset handsets can be very useful. Detailed here is how to set a handset into test mode and what the numbers then mean. Once set go off on a walk round your site to find dead spots. Then change the base position to get the best coverage
The above screen Shows RX power at 100% , Frequency is 3 , TimeSlot is 02, Basestation code is 78 and finally Bit error rate is 100% (This means 100% Good not 100% error rate)
A short document on Setting test mode on Gigaset dect handsets for site surveys is available for download here. This shows you how to enable it and what each of the numbers mean.
The current firmware in the Digium G series gateways have a quirk that if they don’t receive caller ID name they move the caller Id number to be the Caller Id name but don’t leave the Caller Id number in place. The relies on you setting “trustrpid=yes” in teh sip trunk configuration.
We have produced a short document on settings for using the gateway with any freePBX based asterisk solution. It can be downloaded here
Categories
Trusting Linux servers
This hopes to explain in simple steps setting up a pair (or more) servers as a trusted group.
So what do we want to achieve ? Well we wnat to be able to ssh, sftp, rsync etc between servers and not need to enter passwords
Steps required
1 Hosts File
2 Editing sshd_config
3 Create the ssh keys
4 Setting up the Auth. users file
Hosts File
Firstly we need to make sure all servers are in the hosts file
# Do not remove the following line, or various programs
# that require network functionality will fail.
127.0.0.1 localhost
127.0.0.1 asterisk2.local
# We point to eth0 on our own box
192.168.10.100 asterisk2.local
192.168.10.100 asterisk2
# We point to eth1 on the other box
192.168.10.108 asterisk1
Editing sshd_config
Now we need to edit the /etc/ssh/sshd_config file
so that the following
RSAAuthentication yes
PubkeyAuthentication yes
AuthorizedKeysFile /root/.ssh/authorized_keys
replaces
#RSAAuthentication yes
#PubkeyAuthentication yes
#AuthorizedKeysFile .ssh/authorized_keys
Now restart the sshd
/etc/init.d/sshd restart
Create the ssh keys
We now need to create the keys on each server
ssh-keygen -t rsa
and hit return for all the questions.
this will create 2 files in /root/.ssh
go the /root/.ssh directory and copy the id_rsa.pub to the other server and get its id_rsa.pub
sftp asterisk1
put id_rsa.pub asterisk2.pub
get id_rsa.pub asterisk1.pub
bye
Setting up the Auth. users file
In the /root/.ssh directory you will now have for example :-
asterisk1.pub id_rsa id_rsa.pub known_hosts
We now need to copy the asterisk1.pub to the authorized_keys file
cat asterisk1.pub >> authorized_keys
Do the same on the other server.
You should now be able to ssh and rsync between servers.
Categories
Sip debugging with wireshark
Wireshark and Cloudshark are invaluable tools for debugging sip and iax issues on your Asterisk server.
Here we have a short Video that goes over the basics of getting a call captured and opened in Cloudshark
we also have a short tutorial for download here in PDF format
First we need to get the packets we want. This is far simpler than its thought. We use a simple command line tool called tcpdump, if its not installed install it now, You wont be able to live without it.
Here we have 2 commands, The first captures packets on interface eth0, -n means we won’t convert addresses, -w means we just capture raw packets and udp means its only the udp packets we want and finally port 5060 means its only the sip messaging we want. In the second we dont specify port 5060 so that we get the rtp stream as well.
/usr/sbin/tcpdump -n -i eth0 -w /tmp/wireshark.pcap -s2000 udp port 5060 /usr/sbin/tcpdump -n -i eth0 -w /tmp/wireshark.pcap -s2000 udp
screen -S "udpDump" -dm tcpdump -n -i eth0 -C 9 -W 15 -w /var/log/asterisk/dumpsip.pcap -s2000 udp port 5060
The command above will write to file in the background and will rotate at 9 meg so suitable for cloudshark
Once you have started the capture and made a call as required you will get a file called for example /tmp/wireshark.pcap copy this to your workstation via ftp or sftp as you would copy any file.