Over the last few weeks and possibly going on for a few more Gamma Telecom are migrating users from their MSX SBCs to their ‘new’ SWe SBCs, and as side effect of this change is that they now do not support non-symetrical nat translation of RTP traffic
Their previous SBCs and like many other carriers do not have an issue with this and in the words of Twilio’s notes below they support both methods
** When Symmetric RTP is enabled Twilio will detect where the remote RTP stream is coming from and start sending RTP to that destination instead of the one negotiated in the SDP. Please note that this setting is more vulnerable to RTP attacks.
When Symmetric RTP is disabled, Twilio will send RTP to the destination negotiated in the SDP. This setting is considered to be more secure and therefore recommended.
On making support calls to Gamma initially they just seem to tell users that the RTP is being sent from a port that isn’t specified in the SDP, and yes that is correct, But Gamma being Gamma and even though they will have had numerous calls they don’t go any further
It seems the problem is with the customer firewalls in particular pfSense:
By default, pfSense software rewrites the source port on all outgoing connections except for UDP port 500. Some operating systems do a poor job of source port randomization, if they do it at all. This makes IP address spoofing easier and makes it possible to fingerprint hosts behind the firewall from their outbound traffic. Rewriting the source port eliminates these potential (but unlikely) security vulnerabilities. Outbound NAT rules, including the automatic rules, will show in the Static Port column on rules set to randomize the source port.
Source port randomization breaks some rare applications. The default Automatic Outbound NAT ruleset disables source port randomization for UDP 500 because it will almost always be broken by rewriting the source port. Outbound NAT rules which preserve the original source port are called Static Port rules and have on the rule in the Static Port column. All other traffic has the source port rewritten by default.
To add a rule for a device which requires static source ports:
Navigate to Firewall > NAT, Outbound tab
Select Hybrid Outbound NAT rule generation
Click Save
Click to add a new NAT rule to the top of the list
Configure the rule to match the traffic that requires static port, such as a source address of a PBX.
Check Static Port in the Translation section of the page
Click Save
Click Apply Changes
After making that change, the source port on outgoing traffic matching the rule will be preserved. **The best practice is to use strict rules when utilizing static port to avoid any potential conflict if two local hosts use the same source port to talk to the same remote server and port using the same external IP address.**
Personally I would just make this change for the UDP port range and not all UDP ports as this could cause problem with traffic such a port 5060 when multiple servers or phones are on a site.
We have also been made aware of another issue with respect to call diversion to external numbers. By deafault Asterisk and many other IP PBXs set a diversion header in the 181 message giving the device that diverted the call and reason. in most cases this will be the extension number so the header will look like:
This seems to cause issues at Gamma and they reject the call as it seems they are setting the callerid from this info.
To overcome this issue for chan_sip set ‘send_diversion = no’ in the general setting of sip.conf or in the “Other SIP Settings” fields in the Advanced sip setting menu. For PJSIP add it to the pjsip.endpoint_custom_post.conf file as below.
To be honest we have only seen the problem with Gamma trunks and having tested with other suppliers and found they are not affected.
Gammas reson for this is as follows: “After reviewing the divert packet, I can see in the message header that the Diversion header is set to divert to “477”. I would recommend to change this to the full CLI you wish to forward the call to as I believe the system is trying to call “477” which wouldn’t be classed as a valid number. The 603 error you are seeing from your side would be in relation to OFCOMS national number length violation.”
See the Packet below
Session Initiation Protocol (181)
Status-Line: SIP/2.0 181 Call is being forwarded
Status-Code: 181
[Resent Packet: False]
[Request Frame: 22149]
[Response Time (ms): 187]
Message Header
Via: SIP/2.0/UDP xxx.yyy.aaa.zzz:5060;branch=z9hG4bK04B82da620259a59a1a;received=xxx.yyy.aaa.zzz;rport=5060
Transport: UDP
Sent-by Address: xxx.yyy.aaa.zzz
Sent-by port: 5060
Branch: z9hG4bK04B82da620259a59a1a
Received: xxx.yyy.aaa.zzz
RPort: 5060
From: <sip:01234567890@xxx.yyy.aaa.zzz>;tag=gK0441ee4f
SIP from address: sip:01234567890@xxx.yyy.aaa.zzz
SIP from tag: gK0441ee4f
To: <sip:07890123456@aaa.bbb.ccc.ddd>;tag=as24643c1b
SIP to address: sip:07890123456@aaa.bbb.ccc.ddd
SIP to tag: as24643c1b
Call-ID: 71571273_130153708@xxx.yyy.aaa.zzz
[Generated Call-ID: 71571273_130153708@xxx.yyy.aaa.zzz]
CSeq: 321899 INVITE
Sequence Number: 321899
Method: INVITE
Server: FPBX-16.0.40.7(18.9)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:07890123456@aaa.bbb.ccc.ddd:5060>
Contact URI: sip:07890123456@aaa.bbb.ccc.ddd:5060
Contact URI User Part: 07890123456
Contact URI Host Part: aaa.bbb.ccc.ddd
Contact URI Host Port: 5060
Diversion: <sip:477@aaa.bbb.ccc.ddd>;reason=unconditional
Content-Length: 0
Now the RFC says :
“When a diversion occurs, a Diversion header SHOULD be added to the forwarded request or forwarded 3xx response. The Diversion header MUST contain the Request-URI of the request prior to the diversion. The Diversion header SHOULD contain a reason that the diversion occurred.”
Which is what happens, Gamma seem to have confused what the diversion header does as they seem to assume its setting the diversion destination or outbound caller ID, Neither of which are the uses for the Diversion header.
‘I will add updates here as and when they become available.’
It’s the biggest and most important modernisation of the public phone network ever, and your business needs to check and may need to make changes to ensure a smooth transition
In 2017 BT announced it intended to Switch Off ISDN and PSTN by the end of 2025. From September 2023 new ISDN lines will not be available for purchase. Businesses must make alternative plans and migrate all ISDN / PSTN channels or they will be without a telephony service. All equipment that currently uses the PSTN will stop working: such as alarms, elevator phones, EPOS machines, door entry systems etc
There are four options, all suitable for businesses ranging in size from as few as 3 employees to many thousands of employees. All you have to do is decide which is the best fit for you
Option 1: Adapt What You Have
Extend the life of your current phone system by connecting it to the internet. This is simply done by adding hardware known as a VoIP Gateway and a link known as a SIP Trunk, which uses your existing Internet connection. It’s easy, affordable, and users notice no difference – no new cables, no new handsets, no new training.
Option 2: Blend It All Together
Mix options 1, 2, and 3 to suit your needs. For example, an on-premise system at your head office, and a cloud-based system serving your remote sites. Or connect a cloud-based unified communications platform to an on-premise VoIP Gateway or SIP Trunk-powered system. Whatever the blend, enjoy the same seamlessly-integrated user experience.
Option 3: Upgrade What You Have
Replace your installed on-premise system with the latest feature-rich digital technology known as a Unified Communications (UC) Platform; this can be installed on your site as hardware or software, fully under your control. All your telephony now on the internet, but also seamlessly aligned with your email, messaging, and chat applications via an easy-to-use, easily accessible user interface. Plus, it can all be replicated on employees’ desktop computers, laptops and mobile devices for super-convenience.
Option 4: Migrate To The Cloud
Follow hundreds of millions of organisations worldwide by replacing your on-premise system with a powerful, cloud-powered Unified Communications (UC) solution. All your calls, email, chat, and messaging now via the internet; limitless ability to add the latest new features at will; and pay monthly, only for the services you use.
Sangoma have produced a useful Webinar: “How To Prepare For The Great British ISDN Switch Off”Webinar Recording: “How To Prepare For The Great British ISDN Switch Off”
If you have any questions or need advice email or call us.
Before diving into the installation and configuration, it’s better to know some terms used in LDAP.
Attribute
An attribute is a characteristic of an object. For example, an email of an account.
Object Class
An object class defines what attributes that object can have. For example, we define an object class, InetOrgPerson, it may contain displayName and mail attributes. Depends on the definition of object class, the attributes specified can be mandatory or optional.
Distinguished Name (DN)
Distinguished Name lets us uniquely identify the object. It is similar to the file path in a reverse order. For example, uid=JohnDoe,OU=People,DC=abc,DC=local is a DN
Entry
An entry is just an object. You define what object class this entry belongs to & each object class defines what attributes this object has. Each entry can belong to multiple object classes and need to have all mandatory attributes specified in all object classes it belongs to.
Schema
A schema contains the definitions of various attributes and object classes.
Domain Component (DC) & Organizational Unit (OU)
They are containers, contains object & let you manage objects in a hierarchy manner. People use them commonly.
OpenLDAP Installation
Install OpenLDAP related packages
sudo yum install openldap* -y
sudo systemctl start slapd
sudo systemctl enable slapd
sudo systemctl status slapd # Check service is started & enabled
● slapd.service - OpenLDAP Server Daemon
Loaded: loaded (/usr/lib/systemd/system/slapd.service; enabled; vendor preset: disabled)
Active: active (running) since Tue 2023-10-17 11:20:41 BST; 1 weeks 0 days ago
Docs: man:slapd
man:slapd-config
man:slapd-hdb
man:slapd-mdb
file:///usr/share/doc/openldap-servers/guide.html
Main PID: 1922 (slapd)
CGroup: /system.slice/slapd.service
└─1922 /usr/sbin/slapd -u ldap -h ldapi:/// ldap:///
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 fd=22 ACCEPT from IP=192.168.1.202:45777 (IP=0.0.0.0:389)
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 op=0 BIND dn="" method=128
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 op=0 RESULT tag=97 err=0 text=
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 op=1 SRCH base="dc=abc,dc=local" scope=2 deref=0 filter="(|(cn=*)(sn=*))"
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 op=1 SEARCH RESULT tag=101 err=0 nentries=13 text=
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 op=2 UNBIND
Oct 24 16:46:06 testsystem.myserver.co.uk slapd[1922]: conn=1604 fd=22 closed
Oct 24 16:46:49 testsystem.myserver.co.uk slapd[1922]: conn=1530 op=21 SRCH base="dc=abc,dc=local" scope=2 deref=0 filter="(cn=*)"
Oct 24 16:46:49 testsystem.myserver.co.uk slapd[1922]: conn=1530 op=21 SRCH attr=givenName title wWWHomePage telephoneNumber
Oct 24 16:46:49 testsystem.myserver.co.uk slapd[1922]: conn=1530 op=21 SEARCH RESULT tag=101 err=0 nentries=13 text=
OpenLDAP Configuration
Generate OpenLDAP password and save it
sudo slappasswd
Then, we will use ldapmodify to update /etc/openldap/slapd.d/cn=config/olcDatabase={2}hdb.ldif, which is our database config fileWe will create a file & customize and paste content below
vi db.ldif
Content you should paste: You should replace with your customized values
olcSuffix (should be replaced by your domain, e.g. example.com -> dc=example,dc=com)
olcRootDN (should be replaced by your domain admin name, can be any name you prefer, e.g. admin -> cn=admin,dc=abc,dc=local)
olcRootPW (should be the password you generate above)
Apply some commonly used schema. The 2nd & 3rd schema allow us to create an object with InetOrgPerson & ShadowAccount which we will use to create an user
Now the code. I accept no responsibility for it, Its a mess but it does what it says. There is bound to be a better way but with the timescale I had i needed something quick and as such its dirty. each section is distinct so shouldnt be hard to clean up.
The file fpbxldap.sh
#!/bin/bash
#Script file to add delete and modify ladp database for freeepbx contact manager
#Copyright (C) 2023 Ian Plain Cyber-cottage.co.uk
#
#This program is free software; you can redistribute it and/or
#modify it under the terms of the GNU General Public License
#as published by the Free Software Foundation; either version 2
#of the License, or (at your option) any later version.
#
#This program is distributed in the hope that it will be useful,
#but WITHOUT ANY WARRANTY; without even the implied warranty of
#MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
source fldapconfig.sh
# LDAP Entry Details
#BASE_DN="dc=abc,dc=local"
#BASE_OU="ou=People"
# Search for the LDAP entries ad file them
CURRENT_TELEPHONE_NUMBER=$(ldapsearch -x -D "$LDAP_BINDDN" -w "$LDAP_BINDPW" -H "$LDAP_SERVER" -b "$BASE_OU,$BASE_DN" telephoneNumber | awk -F ',|=|: ' '/dn:/ {print $3}')
echo "$CURRENT_TELEPHONE_NUMBER" |grep -w -v "People" > /tmp/ldapdb.txt
# Query to execute
QUERY="SELECT asterisk.contactmanager_entry_numbers.number as 'telephoneNumber', asterisk.contactmanager_group_entries.displayname as 'cn', asterisk.contactmanager_group_entries.fname as 'givenName', COALESCE(NULLIF(asterisk.contactmanager_group_entries.lname, ''), '-') AS 'sn', asterisk.contactmanager_entry_numbers.type as 'o', asterisk.contactmanager_groups.name as 'dir'
FROM asterisk.contactmanager_entry_numbers
INNER JOIN asterisk.contactmanager_group_entries ON asterisk.contactmanager_entry_numbers.entryid=asterisk.contactmanager_group_entries.id
INNER JOIN asterisk.contactmanager_groups ON asterisk.contactmanager_groups.id=asterisk.contactmanager_group_entries.groupid
WHERE asterisk.contactmanager_entry_numbers.number REGEXP '^[0-9]*$' AND asterisk.contactmanager_group_entries.displayname REGEXP '[:alpha:]'
;"
# Output file
OUTPUT_FILE="/tmp/fpbxdb.txt"
# Run the MySQL query and save the result to the output file
mysql -h "$DB_HOST" -u "$DB_USER" -p"$DB_PASSWORD" "$DB_NAME" -N -e "$QUERY" | sed 's/\t/,/g' > "$OUTPUT_FILE"
#Split out just the names
cat /tmp/fpbxdb.txt |awk -F ',' '{print $2" - "$5}' > /tmp/fpxname.txt
cat /tmp/fpbxdb.txt |awk -F ',' '{print $2" - "$5","$1}' > /tmp/fpxnumna.txt
# Assign filenames to variables
listB_file="/tmp/ldapdb.txt"
listA_file="/tmp/fpxname.txt"
# Check if the files exist
if [ ! -f "$listA_file" ]; then
echo "File $listA_file does not exist."
exit 1
fi
if [ ! -f "$listB_file" ]; then
echo "File $listB_file does not exist."
exit 1
fi
#Bit of a hack here that adds an entry to empty file, as AWK doesnt like empty files.. Thsi was quick a fix
if [ -s "$listA_file" ]; then
echo "The file is not empty."
else
echo "foobar" > /tmp/fpxname.txt
fi
if [ -s "$listB_file" ]; then
echo "The file is not empty."
else
echo "barfoo" > /tmp/ldapdb.txt
fi
# Compare the two files and echo names in List A but not in List B
awk 'NR==FNR{a[$0]++; next} !a[$0]' "$listB_file" "$listA_file" > /tmp/add.txt
awk 'NR==FNR{a[$0]++; next} !a[$0]' "$listA_file" "$listB_file" > /tmp/rem.txt
#lets delete the entries
# Loop through each line in the input file and run the command
while IFS= read -r REM_FILTER; do
# Run the specified command on each line
echo "$REM_FILTER deleted from Ldap" >> /tmp/remlog.txt
ldapdelete -x -D "$LDAP_BINDDN" -w "$LDAP_BINDPW" -H "$LDAP_SERVER" "cn=$REM_FILTER,$BASE_OU,$BASE_DN"
done < "$rem_file"
echo "Done-------" >> /tmp/remlog.txt
#delete the previous ldif files
rm -f /tmp/adding.ldif
rm -f /tmp/modify.ldif
#lets add the entries
# Loop through each line in the input file and run the command
while IFS= read -r ADD_FILTER; do
# Run the specified command on each line
# echo $ADD_FILTER |awk -F ' - ' '{print $1; print $2}'
ms_cn="$(echo $ADD_FILTER |awk -F ' - ' '{print $1}')"
ms_o="$(echo $ADD_FILTER |awk -F ' - ' '{print $2}')"
# Query to execute
QUERY="SELECT asterisk.contactmanager_entry_numbers.number as 'telephoneNumber', COALESCE(NULLIF(asterisk.contactmanager_group_entries.lname, ''), '-') AS 'sn'
FROM asterisk.contactmanager_entry_numbers
INNER JOIN asterisk.contactmanager_group_entries ON asterisk.contactmanager_entry_numbers.entryid=asterisk.contactmanager_group_entries.id
INNER JOIN asterisk.contactmanager_groups ON asterisk.contactmanager_groups.id=asterisk.contactmanager_group_entries.groupid
WHERE asterisk.contactmanager_entry_numbers.type = '$ms_o' AND asterisk.contactmanager_group_entries.displayname = '$ms_cn'
;"
# Run the MySQL query and save the result to the output file
ms_query=$(mysql -h "$DB_HOST" -u "$DB_USER" -p"$DB_PASSWORD" "$DB_NAME" -N -e "$QUERY")
ms_telephoneNumber=$(echo $ms_query | awk '{print $1}')
ms_sn=$(echo $ms_query | awk '{print $2}')
echo "dn: cn=$ms_cn - $ms_o,ou=People,dc=abc,dc=local" >> /tmp/adding.ldif
echo "cn: $ms_cn - $ms_o" >> /tmp/adding.ldif
echo "givenName: $ms_cn - $ms_o" >> /tmp/adding.ldif
echo "sn: $ms_sn" >> /tmp/adding.ldif
echo "telephonenumber: $ms_telephoneNumber" >> /tmp/adding.ldif
echo "objectclass: inetOrgPerson" >> /tmp/adding.ldif
echo "objectclass: top" >> /tmp/adding.ldif
echo "" >> /tmp/adding.ldif
echo "cn: $ms_cn - $ms_o , $ms_telephoneNumber added to Ldap" >> /tmp/addlog.txt
done < "$add_file"
#Lets run the ldif command
ldapadd -x -D "$LDAP_BINDDN" -w "$LDAP_BINDPW" -H "$LDAP_SERVER" -f /tmp/adding.ldif >> /tmp/addlog.txt
echo "Done-------" >> /tmp/addlog.txt
#OK now we are going to compare freepbx and ldap entries and update as required.
#lets get the current ldap names and numbers
ldapsearch -x -D "cn=admin,dc=abc,dc=local" -w "r1v3rp1g5" -b "ou=People,dc=abc,dc=local" | awk -v OFS=',' '{split($0,a,": ")} /^cn:/{cn=a[2]} /^telephoneNumber:/{telephoneNumber=a[2]; print cn,telephoneNumber}' > /tmp/ldapcsv.txt
awk 'NR==FNR{a[$0]++; next} !a[$0]' /tmp/ldapcsv.txt /tmp/fpxnumna.txt > /tmp/chg.txt
chg_file="/tmp/chg.txt"
# Loop through each line in the input file and run the command
while IFS= read -r CHG_FILTER; do
# Run the specified command on each line
# echo $CHG_FILTER |awk -F ',' '{print $1; print $2}'
ms_cn="$(echo $CHG_FILTER |awk -F ',' '{print $1}')"
ms_telephoneNumber="$(echo $CHG_FILTER |awk -F ',' '{print $2}')"
echo "Changing telephoneNumber to $ms_telephoneNumber"
echo "dn: cn=$ms_cn,$BASE_OU,$BASE_DN" >> /tmp/modify.ldif
echo "changetype: modify" >> /tmp/modify.ldif
echo "replace: telephoneNumber" >> /tmp/modify.ldif
echo "telephoneNumber: $ms_telephoneNumber" >> /tmp/modify.ldif
echo "" >> /tmp/modify.ldif
echo "$CHG_FILTER changed in Ldap" >> /tmp/chglog.txt
done < "$chg_file"
ldapmodify -x -D "$LDAP_BINDDN" -w "$LDAP_BINDPW" -H "$LDAP_SERVER" -f /tmp/modify.ldif >> /tmp/modify.ldif
echo "Done-------" >> /tmp/chglog.txt
Example phone configurations for Sangoma S series and Gigaset.
Gigaset example
Sangoma S Series example
Im sure this will work with other systems that support Ldap directories
For this project we are going to use the Amazon AWS Transcribe service, AWS Transcribe is a cloud-based speech recognition service that converts audio recordings into accurate text transcripts. It uses advanced machine learning algorithms to identify different speakers and punctuation, while also supporting a variety of audio formats and languages. AWS Transcribe can transcribe audio from sources such as phone calls, video recordings, and live streams, making it a versatile tool thats idealy suited for voicemail transcription, The service is highly scalable and cost-effective.
We will say that we used to use Google’s Text to speech engine for thsi but over time I would have expected quality of transcription to have improved, But with Google this is not the case, and I expect this is because they possibly use “predictive” text to speech and not sample all the words as this example below shows, This is the same audio fed to Google and AWS
Amazon AWS Transcribe
Um, this is Ian. I’d like to order some pizza for tomorrow, please. We would like to order a pepperoni pizza and a mozzarella pizza that’s for tomorrow at five PM. Thank you.
Google Speech to Text
like to order some pizza for tomorrow please would like to order a pepperoni pizza and a mozzarella Pizza Hut for tomorrow at 5 a.m. thank you
As can be seen google misses words and adds others, As you can imagine this isnt what you want with speech transcription.
So we have switched out old script to use AWS.
For this project on Freepbx you need a few extra applications added and a amazon aws account, setting this up is not covered here as you should already have knowledge of this if you are here.
The extra apps are , aws , jq , sox
to get aws :
curl "https://awscli.amazonaws.com/awscli-exe-linux-x86_64.zip" -o "awscliv2.zip"
unzip -qq awscliv2.zip
./aws/install
Then you need to configure as 'root' and as 'asterisk', so:
aws configure
fill out your aws key and token as well as the region your bucket is in
Then repeat as 'asterisk' so
su asteriskaws configure
and fill out same details.
for jq and sox, just yum install xxx as you would for any other program.
Next you need the asterisk dialplan added to the extensions_custom.conf
as can be seen this dialplan records a file and then runs the vmailprox.sh script. This script collects the variables and passes them over to the main script and exits after doing so, this is so channels aren’t held while transcription takes place. (Thats the plan anyway)
#!/bin/sh
PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin"
S3_BUCKET="YOURS3BUCKET"
DIRPATH=/var/spool/asterisk/voicemail/default/
#callerchan=$1
#callerid=$2
#origdate=$3
#origtime=$4
#origmailbox=$5
#origdir=$6
#duration=$7
counter=1
sleep 4
FILENUM=$(/bin/ls ${DIRPATH}${origmailbox}/INBOX |/bin/grep txt | /usr/bin/wc -l)
##Added to allow 999 messages
if (( $FILENUM <= 9 ));
then
FILENAME=msg000${FILENUM}
elif (( $FILENUM <= 99 ));
then
FILENAME=msg00${FILENUM}
else
FILENAME=msg0${FILENUM}
fi
IN=$(/bin/grep "${origmailbox}=" /etc/asterisk/voicemail.conf)
set -- "$IN"
IFS=","; declare -a Array=($*)
email=${Array[2]}
/bin/echo "[message]" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo origmailbox=${origmailbox} >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "context=demo" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "macrocontext=" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "exten=s" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "priority=11" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo callerchan=${callerchan} >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo callerid=${callerid} >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo origdate=${origdate} >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo origtime=${origtime} >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo msg_id=${origtime}-00000001 >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "flag=" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "category=" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/echo "duration=${duration}" >> ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.txt
/bin/nice /usr/bin/lame -b 16 -m m -q 9-resample /var/lib/asterisk/sounds/catline/${origdir}.wav /tmp/${origdir}.mp3
# Create a string based on the current date and time
current_date_time="$(date +%Y-%m-%d_%H-%M-%S)"
# Upload to the S3 Bucket
aws --debug --profile default s3 cp /tmp/${origdir}.mp3 s3://$S3_BUCKET/$current_date_time
# Start the transcription job
output=$(aws --profile default transcribe start-transcription-job \
--transcription-job-name $current_date_time \
--language-code en-GB \
--media-format mp3 \
--media MediaFileUri=s3://$S3_BUCKET/$current_date_time \
--output-bucket-name $S3_BUCKET)
# Wait for the transcription to finish
JOB_COMPLETED=false
while [ "$JOB_COMPLETED" = false ]; do
JOB_STATUS=$(aws --profile default transcribe get-transcription-job \
--transcription-job-name $current_date_time \
--query 'TranscriptionJob.TranscriptionJobStatus' \
--output text)
if [ "$JOB_STATUS" = "FAILED" ]; then
JOB_COMPLETED=true
SHORT_CALL=yes
/bin/echo "$JOB_STATUS" >> /tmp/logfile.txt
break
fi
if [ "$JOB_STATUS" = "COMPLETED" ]; then
/bin/echo "$JOB_STATUS" >> /tmp/logfile.txt
JOB_COMPLETED=true
else
((counter++))
sleep 5
echo $counter >> /tmp/logfile.txt
/bin/echo "$JOB_STATUS" >> /tmp/logfile.txt
if [ "$counter" -eq "15" ]; then
JOB_STAUS=COMPLETED
JOB_COMPLETED=true
SHORT_CALL=yes
break
fi
fi
done
# Get the transcription result
aws s3 --profile default cp s3://$S3_BUCKET/$current_date_time.json /tmp/$current_date_time.json
# Get the transcription result
FILTERED=$(jq -r '.results.transcripts[].transcript' /tmp/$current_date_time.json)
# append result of transcription
if [ -z "$FILTERED" ]
then
echo "(AWS was unable to recognize any speech in audio data.)" >> /tmp/${origdir}.txt
else
echo "$FILTERED" >> /tmp/${origdir}.txt
sed -i 's/ Um,/ /gI' /tmp/${origdir}.txt
fi
voicemailbody=$(cat "/tmp/${origdir}.txt")
# echo "body ${voicemailbody}"
/bin/cp /var/lib/asterisk/sounds/catline/${origdir}.wav ${DIRPATH}${origmailbox}/INBOX/${FILENAME}.wav
echo -e "You have a new voicemail from ${callerid} it was left on ${origdate} and is ${duration} seconds long,\nThe message left,\n\n${voicemailbody}\n\nTranscribed by the Amazon AWS Transcribe service\n" | /bin/mail -s "A new voicemail has arrived from ${callerid}" -a "/tmp/${origdir}.mp3" "$email"
/bin/rm -f /tmp/${origdir}.mp3
/bin/rm -f /tmp/${origdir}.txt
aws --profile default transcribe delete-transcription-job --transcription-job-name $current_date_time
Then to pass calls to this and not normal voicemail, In Freepbx create a Custom Destination as “vmail2text,s,1” and if you require certain queues to go to specific mailboxes for example 2000 one like “vmail2text,2000,1” so calls will be sent to mailbox 2000 and teh transcriptions will be sent to the email address linked to that extension
Then in extensions that want to use transcription set the “Optional Destinations” in the advanced tab to the custom destination.
Users also can listen to voicemail normally from their handset or the ucp.
These scripts arent only useful for voicemail then can be used fro questionnaire lines and booking lines, anywhere you want to speed up the handling of voice messages. We will soon be looking at ways of integrating this with Whatsapp so transcriptions can be sent to your mobile.
Sangoma Desktop Softphone turns a user’s computer into a fully featured phone that is primed to take full advantage of all of Sangoma’s unified communications features.
Sangoma Desktop Softphone For Desktop turns a user’s computer into a fully featured phone that is primed to take full advantage of all of Star2Star’s unified communications features. Sangoma Desktop Softphone lets users make, answer, hold, and transfer calls, participate in voice conferencing, access voicemail, integrate calls with CRM software, and more.
Features
Workforce Flexibility: Your staff will no longer be tethered to one location, allowing employees to work remotely while expanding your potential pool of employees to a much greater geographic area.
Cost Savings: Instead of purchasing a computer and a desk phone, you can just purchase a computer and a headset, saving hundreds or even thousands of dollars with softphone software.
Efficiency: Employees can stay in contact on the go or from home, softphones allow them to respond faster. Even better, Desktop Softphone allows them to take advantage of our many powerful features that will go anywhere they go, unlike a traditional mobile phone that lacks much of this functionality.
Incredible Functionality: With Desktop Softphone, your employees have access to a complete unified communications system at their keyboard, meaning they can do it all wherever they are.
Business Voice & Business Voice+ Compatible: Use Desktop Softphone with either of the Full Spectrum Communications platforms for the ultimate flexibility.
Replacing the R650H PRO, the R700H PRO is a robust, IP65 rated Gigaset professional handset. Designed to protect against impacts, water and dust as well as being resistant to disinfectants, the R700H offers users a business ready handset no matter the conditions.
With a 2.4” colour display and Bluetooth headset connection, the R700H is an enduring companion with an extended battery life of up to 13 hours of talk time and 320 hours on standby. What’s more the R700H comes with additional unique features like the spot LED for call signalling or to be used as a torch, as well as a separately programmable alarm button to easily trigger a direct dial or alarm call.
The R700H is fully compatible with the professional Gigaset DECT single and multi-cell systems, including the N670IP , N510IP , N870IP, N720IP.
IP65 and disinfectant resistant In addition to the rubberized surface which gives the device a perfect grip the new R700H PRO is designed in such a way that it can also be operated with work gloves. The display glass is proof against scratches thanks to a harder surface and can therefore be used in extreme working environments. With an IP65 certification, the R700H PRO is protected against shocks and against the ingress of dust and water jets. In addition, the special magnetic shielding makes the R700H PRO resistant to metal dust. The handset has a disinfectant-resistant surface, which simplifies cleaning and thus protects against bacteria and virucides.
Feature highlights
Shockproof, dustproof and waterproof according to IP65
Non-slip rubberised surface
Programmable alarm-/function button
Large illuminated 2.4″ TFT colour display
Headset operation via Bluetooth 4.2 or 3.5mm jack
Handsfree talking with brilliant HDSP™/CAT-IQ 2.01
acoustic quality and high maximum volume
Spot LED as torch and for call signalling
Hotel option: call lists are automatically cleared
Audio profiles for quick selection in a meeting, in a loud
environment or with individual settings
SUOTA: software update over the air
Disinfectant resistant and scratch proof surface
Local phone book with search function and up to 500 vCards
and access to company phone book via PBX (XML, LDAP)3
Data exchange via Bluetooth® or Micro-USB
Vibrating alert
No ringing in the charging cradle can be set for parallel calls
Key lock with PIN protection – emergency call dialling despite
PIN protection
Up to 13h talk time
Up to 320h standby time
Charging also via Micro-USB connection
Charging cradle included free of charge
Full compatibility with the professional Gigaset DECT single
and multi-cell systems
Email or Call for current pricing and qty discounts
Combining the slim-line design of a smartphone with the technology of a professional DECT mobile device, the Gigaset SL800H PRO offers users the best experience in DECT mobile devices. With a lightweight design, large 2.4” TFT colour display as well as advanced sound quality and long battery life, the SL800H PRO offers the highest level of mobility freedom including connectivity to headsets or Bluetooth.
The SL800H PRO offers users all day usage with its extended battery life of up to 15 hours of talk time with a range of up to 50 metres indoors & 300 metres outdoors. Users can rest assured not to miss calls with vibrating alert as well as individual ringer melodies for VIP entries and internal callers.
What’s more the SL800H PRO offers users the option of using the SL800H PRO as a Hotel phone, by reducing options linking to internal business functions such as reduced calendar functions and Bluetooth menu but allowing for quick data exchange and deleting call lists on a time-controlled basis.
This handset is fully compatible with both Gigaset N510IP PRO and N670IP singlecell solutions as well as the N720IP PRO and N870IP PRO multicell DECT solution.
Feature highlights • Large illuminated 2.4″ TFT colour display • Headset operation via Bluetooth® 4.2 or 3.5mm jack • Up to 15h talk time and 300h standby time • Audio profiles can be selected quickly using a separate key • Handsfree with brilliant HDSP ready TM/CAT -IQ 2.01 Acoustic quality and high maximum volume • Hotel option – automatic deletion of sensitive data • SUOTA – Software update via the air interface • Scratch and disinfectant resistant • Local telephone book with search function and up to 500 vCards and access to the company telephone book via PBX (XML, LDAP)2 • Data exchange via Bluetooth® or Micro-USB • Vibrating alarm • No ringing in the charging cradle adjustable for parallel call • Key lock with PIN protection • Charging: • Charging also via Micro-USB connection • Charging cradle included free of charge • Full compatibility with the professional Gigaset DECT Single and multi-cell systems
Email or Call for current pricing and qty discounts
The S700H is the perfect all round DECT handset for daily use. Whether that be in a professional office environment or a busy hotel, the S700H stands out from the rest. With a large 2.4” TFT display that fits comfortably in hand, the S700H offers high quality audio in a scratch and disinfectant resistant IP40 protective casing.
With a separately programmable alarm-button, built-in Bluetooth 4.2 and integrated 3.5mm jack connection, the S700H offers a multitude of additional features and functions. Not to mention an extended talk time of up to 13 hours.
The S700H is available as a hotel option. Allowing for more control over the features available on the S700H. Including blocking soft key programming, deactivated Bluetooth menu, voicemail deactivation and time-controlled data deletion such as call lists.
Feature highlights • Large illuminated 2.4″ TFT colour display • Intuitive user interface for easy operation • Headset operation via Bluetooth® 4.2 or 3.5mm jack • Separately programmable alarm button • Audio profiles can be selected quickly using a separate key • Side buttons for volume adjustment during the Conversation • IP40 protection class and tightness against metal dust • Handsfree with brilliant HDSP ready TM/CAT -IQ 2.01 Acoustic quality and high maximum volume • Hotel option – automatic deletion of sensitive data • SUOTA – Software update via the air interface • Scratch and disinfectant resistant • Local telephone book with search function and up to 500 vCards and access to the company telephone book via PBX (XML, LDAP)2 • Data exchange via Bluetooth® or Micro-USB • Vibrating alarm • No ringing in the charging cradle adjustable for parallel call • Key lock with PIN protection • Charging: • Up to 12h talk time • Up to 300h standby time • Charging also via Micro-USB connection • Charging cradle included free of charge • Full compatibility with the professional Gigaset DECT Single and multi-cell systems
Email or Call for current pricing and qty discounts
Sangoma’s line of P-Series phones are designed to deliver the features you need, at price points perfect for every type of user, and every type of business.
All models include color screens, high definition voice, are headset-ready, provide unprecedented plug-and-play deployment, and have built-in productivity applications including voicemail, call log, contacts, phone status, user presence, parking and more.
Sangoma’s P-Series phones are the only phones that are compatible across Sangoma’s communications as a service portfolio, further enhancing their value.
Entry-Level – The Sangoma P310/P315 Phones Sangoma’s value-line is perfect for large floor deployments in offices, schools, manufacturing, and retail.Mid-Range – The Sangoma P320, P325, P330 Phones Perfect for knowledge workers, with built-in business applications*, controlled via a large 4.3-inch IPS display and plenty of programmable function keys. The P330 also supports built-in Bluetooth and WiFi as well as the forthcoming PM200 expansion module.Executive – The Sangoma P370 Phone (coming soon!) For the executives in the office who demand a sleek desktop presence, the P370 delivers. With a large 7.0” 1280×800 color IPS touchscreen display, built-in Bluetooth and WiFi, all of Sangoma’s business applications* are easy-to-use with finger-touch access.
Email or Call for current pricing and qty discounts
Since this post was originally written things have advanced, FreePBX has an integrated firewall with intrusion detection using Fail2Ban, and this should always be enabled even if system is on premise.
Another major step forward in protection is APIBAN this is a client program that helps prevent unwanted SIP traffic by identifying addresses of known bad actors before they attack your system. Bad bots are collected through globally deployed honeypots. To use APIBAN you will need a key these are obtained from here . More details on API ban are here if you are interested in using it in different situations.
To simplify installation on Freepbx based systems I have simple script that downloads and install it, this can be downloaded here or from the command line of the server as follows:
wget https://freeaccesspublic.s3.eu-west-2.amazonaws.com/apiban.sh
Make it an executable : chmod +x apiban.sh
then run the script : ./apiban.sh your_api_key
If you dont add your APIKEY on the command line vi will open and you can add it manually. The script will then initially run the client which will take a few seconds to download the initial set of bots, then it will add a line to the crontab file and restart the cron daemon. the timing of the cronjob is randomised to be between every 4 and 22 minutes.
We have seen many Bots attacking Asterisk servers, Interestingly its not always good old sipvicious anymore but a Windows program called sipcli and originating mainly from the US and Germany.
Normally our iptables firewalls are updated but for some reason these keep getting through, So we have now based rules on the User-Agent in iptables as well
Here are a few examples to get rid of many of the favourites
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: friendly-scanner" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: sipcli" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: VaxSIPUserAgent" --algo bm --to 65535 -j DROP
For Freepbx format add following to the Firewalls custom rules
-A fpbxreject -p udp --dport 5060:5261 -m string --string "REGISTER sip:server.domain.co.uk" --algo bm -j ACCEPT
-A fpbxreject -p udp --dport 5060:5261 -m string --string "REGISTER sip:" --algo bm -j DROP
-A fpbxreject -p tcp --dport 5060:5261 -m string --string "REGISTER sip:server.domain.co.uk" --algo bm -j ACCEPT
-A fpbxreject -p tcp --dport 5060:5261 -m string --string "REGISTER sip:" --algo bm -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "sip:a'or'3=3--@" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: PolycomSoundPointIP SPIP_550 UA 3.3.2.0413" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Avaya IP Phone 1120E" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: PolycomVVX-VVX_401-UA5.4.1.18405" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: eyeBeam release 3006o stamp 17551" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: owenee" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: owenee" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Custom" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: Custom" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: SIP" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: SIP" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: gazllove" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: gazllove" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: pplsip" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: pplsip" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipcli" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipcli" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sip-scan" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sip-scan" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipsak" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipsak" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sundayddr" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sundayddr" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: friendly-scanner" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: friendly-scanner" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: iWar" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: iWar" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: CSipSimple" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: CSipSimple" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: SIVuS" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: SIVuS" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Gulp" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: Gulp" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipv" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipv" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: smap" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: smap" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: friendly-request" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: friendly-request" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: VaxIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: VaxIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: VaxSIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: VaxSIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: siparmyknife" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: siparmyknife" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Test" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: Test" --algo bm --to 65535 -j DROP
Also its worth adding these ranges as little good will ever come from them
# Ponytelecom ranges
-A INPUT -s 62.210.0.0/16 -j DROP
-A INPUT -s 195.154.0.0/16 -j DROP
-A INPUT -s 212.129.0.0/18 -j DROP
-A INPUT -s 62.4.0.0/19 -j DROP
-A INPUT -s 212.83.128.0/19 -j DROP
-A INPUT -s 212.83.160.0/19 -j DROP
-A INPUT -s 212.47.224.0/19 -j DROP
-A INPUT -s 163.172.0.0/16 -j DROP
-A INPUT -s 51.15.0.0/16 -j DROP
-A INPUT -s 151.115.0.0/16 -j DROP
# VITOX TELECOM
-A INPUT -s 77.247.109.0/255.255.255.0 -p udp -j DROP
-A INPUT -s 185.53.88.0/24 -p udp -j DROP
-A INPUT -s 185.53.89.0/24 -p udp -j DROP
-A INPUT -s 37.49.224.0/24 -p udp -j DROP
-A INPUT -s 37.49.230.0/24 -p udp -j DROP
-A INPUT -s 37.49.231.0/24 -p udp -j DROP
-A INPUT -s 77.247.110.0/255.255.255.0 -p udp -j DROP
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