Category: Knowledge Base

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Elastix Support Software Releases

Elastix 2.4 Released

Elastix.org have announced the release of 2.4 stable.elastix240_en

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
Knowledge Base

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

Explanation of Terms

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
Categories
Knowledge Base Support

24×7 Asterisk server monitoring with Nagios.

We offer an economical solution for end users and resellers to monitor their Asterisk and Linux servers.

Our platform monitors servers 24 hours a day 7 days a week. Hosted in a state of the art US based data centre with connections to major UK data centres and multiple connections to the internet.

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. 

Nagios monitor screen

 

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 

Categories
FIrmware releases

Gigaset N300 IP, N300A IP, N510 IP PRO – Firmware update 12/2012 (version 075) released

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
Categories
FIrmware releases Knowledge Base

Yealink release V70 firmware for their T2X Sets

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.

Download release notes here

 

Categories
Knowledge Base

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.

Categories
Knowledge Base

Digium G100/200 Gateways and UK CallerID Number

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
Knowledge Base

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
Knowledge Base

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.

Categories
Asterisk Support Elastix Support Knowledge Base

Backing up to Amazon s3 from Elastix

We decided to do this as we have recently installed a new elastix server in the office which had limited disk space and wanted to keep offsite backups of recordings

s3cmd is a command line client for copying files to/from Amazon S3 (Simple Storage Service) and performing other related tasks, for instance creating and removing buckets, listing objects, etc.

Install s3cmd

wget http://sourceforge.net/projects/s3tools/files/s3cmd/1.5.2/s3cmd-1.5.2.tar.gz
 tar -xzvf s3cmd-1.1.0-beta3.tar.gz
 mkdir /usr/local/s3cmd/
 cd s3cmd-1.1.0-beta3
 cp -Rf * /usr/local/s3cmd/
 cd /usr/local/s3cmd/
 ./s3cmd --configure

Follow the prompts and enter your keys.

Test the installation
./s3cmd ls s3://yourbucket/

If the test works then the script below is a simple backup script to backup elastix monitor files and backups daily.

vi /etc/cron.daily/rec2s3c.sh

#!/bin/sh
 /usr/local/s3cmd/s3cmd --config=/some/where/.s3cfg sync /var/spool/asterisk/monitor s3://yourbucket
 /bin/rm -f /var/spool/asterisk/monitor/*.gsm
 /bin/rm -f /var/spool/asterisk/monitor/*.wav
 /usr/local/s3cmd/s3cmd --config=/some/where/.s3cfg ls s3://yourbucket/monitor/ > /var/log/s3dirlist.log
 /usr/local/s3cmd/s3cmd --config=/some/where/.s3cfg sync /var/www/backup s3://yourbucket
 /usr/local/s3cmd/s3cmd --config=/some/where/.s3cfg ls s3://yourbucket/backup/ >> /var/log/s3dirlist.log

 

enjoy :-)

For more details of what can be done with s3cmd see http://linux.die.net/man/1/s3cmd and http://aws.amazon.com/s3/