Category: Technical

Categories
Knowledge Base

ISDN alarms and what they mean.

AIS (Alarm Indication Signal) CFA
The AIS is also known as a “Keep Alive” or “Blue Alarm” signal. This consists of an UNFRAMED all-ones signal sent to maintain transmission continuity. The AIS CFA signal is declared when both the AIS state and RED CFA persist simultaneously.


OOF (Out-Of-Frame) Condition
Occurs whenever Network or DTE equipment senses errors in the incoming framing pattern. Depending upon the equipment, this can occur when 2 of 4, 2 of 5, or 3 of 5 framing bits are in error. A reframe clears the OOF condition.
Red CFA (Carrier Failure Alarm)
Occurs after detection of CONTINUOUS OOF condition for 2.5 seconds. This alarm state is cleared when no OOF conditions occur for AT LEAST 1 second. Some applications (AT&T DACS services) may not clear the CFA state for UP TO 15 seconds of NO Out-Of-Frame occurrences.


Yellow CFA (Carrier Failure Alarm)
When a Terminal/Network equipment enters a RED CFA state, it transmits a “Yellow Alarm” in the opposite direction.
A Yellow Alarm is transmitted by setting Bit #2 of each timeslot to a 0 (zero), SPACE state for D4 Framed facilities. For ESF facilities, a Yellow Alarm is transmitted by sending a repetitive 16-bit pattern consisting of 8 MARKS (1) followed by 8 SPACES (0) in the Datalink bits. This is transmitted for a MINIMUM of 1 second.
LOS (Loss Of Signal)
A LOS condition is declared when no pulses have been detected in a 175 +/- 75 pulse window (100 to 250 bit times).

Notes taken from Digiums Realease notes

Alarm Types

An alarm indicates that a port is not available for some reason. Thus it is probably not a good idea to try to call out through it.

Red Alarm

Your T1/E1 port will go into red alarm when it cannot maintain synchronization with the remote switch. A red alarm typically indicates either a physical wiring problem, loss of connectivity, or a framing and/or line-coding mismatch with the remote switch.

When your T1/E1 port loses sync, it will transmit a yellow alarm to the remote switch to indicate that it’s having a problem receiving signal from the remote switch.

The easy way to remember this is that the R in red stands for “right here” and “receive”… indicating that we’re having a problem right here receiving the signal from the remote switch.

Yellow Alarm

(RAI — Remote Alarm Indication)

Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to
maintain sync with you, or is not receiving your transmission.

The easy way to remember this is that the Y in yellow stands for “yonder”… indicating that the remote switch (over yonder) isn’t able to see what you’re sending.

Blue Alarm

(AIS — Alarm Indication Signal)

Your T1/E1 port will go into blue alarm when it receives all unframed 1s on all timeslots from the remote switch. This is a special signal to indicate that the remote switch is having problems with its
upstream connection. dahdi_tool and Asterisk don’t correctly indicate a blue alarm at this time. The easy way to remember this is that streams are blue, so a blue alarm indicates a problem upstream from
the switch you’re connected to.

Categories
Knowledge Base

Better SIP security

In Seven Steps

Original Text by J Todd March 28th, 2009

In case any of you were wondering why there has been a fairly notable upswing in the attacks happening on SIP endpoints, the answer is “script kiddies.”  In the last few months, a number of new tools have made it easy for knuckle-draggers to attack and defraud SIP endpoints, Asterisk-based systems included.  There are easily-available tools that scan networks looking for SIP hosts, and then scan hosts looking for valid extensions, and then scan valid extensions looking for passwords.You can take steps, NOW, to eliminate many of these problems.  I think the community is interested in coming up with an integrated Asterisk-based solution that is much wider in scope for dynamic protection (community-shared blacklists is the current thinking) but that doesn’t mean you should wait for some new tool to defend your systems.  You can IMMEDIATELY take fairly common-sense measures to protect your Asterisk server from the bulk of the scans and attacks that are on the increase. The methods and tools for protection already exists – just apply them, and you’ll be able to sleep more soundly at night.

Seven Easy Steps to Better SIP Security on Asterisk:

 

1) Don’t accept SIP authentication requests from all IP addresses.  Use the “permit=” and “deny=” lines in sip.conf to only allow a reasonable subset of IP addresess to reach each listed extension/user in your sip.conf file.  Even if you accept inbound calls from “anywhere” (via [default]) don’t let those users reach authenticated elements!

 

2) Set “alwaysauthreject=yes” in your sip.conf file.  This option has been around for a while (since 1.2?) but the default is “no”, which allows extension information leakage.  Setting this to “yes” will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks.

 

3) Use STRONG passwords for SIP entities.  This is probably the most important step you can take.  Don’t just concatenate two words together and suffix it with “1? – if you’ve seen how sophisticated the tools are that guess passwords, you’d understand that trivial obfuscation like that is a minor hinderance to a modern CPU.  Use symbols, numbers, and a mix of upper and lowercase letters at least 12 digits long.

 

4) Block your AMI manager ports.  Use “permit=” and “deny=” lines in manager.conf to reduce inbound connections to known hosts only.  Use strong passwords here, again at least 12 characters with a complex mix of symbols, numbers, and letters.

 

5) Allow only one or two calls at a time per SIP entity, where possible.  At the worst, limiting your exposure to toll fraud is a wise thing to do.  This also limits your exposure when legitimate password holders on your system lose control of their passphrase – writing it on the bottom of the SIP phone, for instance, which I’ve seen.

 

6) Make your SIP usernames different than your extensions.  While it is convenient to have extension “1234? map to SIP entry “1234? which is also SIP user “1234?, this is an easy target for attackers to guess SIP authentication names.  Use the MAC address of the device, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, try “md5 -s ThePassword5000?)

 

7) Ensure your [default] context is secure.  Don’t allow unauthenticated callers to reach any contexts that allow toll calls.  Permit only a limited number of active calls through your default context (use the “GROUP” function as a counter.)  Prohibit unauthenticated calls entirely (if you don’t want them) by setting “allowguest=no” in the [general] part of sip.conf.

 

These 7 basics will protect most people, but there are certainly other steps you can take that are more complex and reactive.  Here is a fail2ban recipe which might allow you to ban endpoints based on volume of requests.  There is discussion on the asterisk-user and asterisk-dev mailing lists of incorporating this type of functionality into Asterisk – let’s hear your ideas!

 

If you’d like to see an example of the tools that you’re up against, see this demo video of an automated attack tool that does scan, guess, and crack methods via a click-and-drool interface.
In summary: basic security measures will protect you against the vast majority of SIP-based brute-force attacks.  Most of the SIP attackers are fools with tools – they are opportunists who see an easy way to defraud people who have not considered the costs of insecure methods.  Asterisk has some methods to prevent the most obvious attacks from succeeding at the network level, but the most effective method of protection are the administrative issues of password robustness and username obscurity.

 

JTodd
Digium
Categories
Asterisk Support Elastix Support Knowledge Base OpenVox

Asterisk pickup groups

The aim here is to explain the relationship between the callgroup and pickup group settings in extension conf files of an Asterisk server and named pickup in freepbx, we will use numbers but not names (see explanation below).

Call Pickup is the abilty to pickup a ringing phone from another phone.

The ability to do this is defined in the extensions conf file.

In many systems there is only on setting to do this normally “pickup group” you add extensions to this group and they can pickup calls ringing at members of the group. Obvious really.

Now Asterisk goes one better. You can define the callgroup and pickup group, This way you define who you can pickup and who can pickup you. This is very useful for operators, who for example don’t want calls picked up of them but do want to pickup calls from all other users.

So how do you define it.

In our example we will have 4 phones defined as follows

Callgroup Pickupgroup
201 2 1-2
202 1-4 1-4
203 2,4 2,4
204 1 1

And who can do what when trying t pickup is as follows

Ringing Phones attempting Pickup
Call to 201 204 PU failed 203 PU Passed
Call to 202 201 PU passed 203 PU Passed
Call to 203 201 PU passed 204 PU failed
Call to 204 201 PU passed 203 PU failed

So from this we can see that its the Pickupgroup that defines what callgroup can be picked up.

So because 201 has a callgroup of 2 Only sets who’s pickup group includes 2 can pick up the call. whereas as 201 has a pickupgroup of 1-2 it can pickup calls from callgroups 1-2.

For example you may have 6 pickup groups defined with users only allowed to pickup their own group members except an operato who wishes to be able to pick everyone up and a PA who has a college who she wants to be able to pickup

So all normal users would have their pickup and callgroup the same. The PA would have the pickupgroup defined with both the group numbers but only its own call group. And finally the operator would have a callgroup of 0 and its pickupgroup of 1-6.

Named call pickup groups

Named pickup groups are new with Asterisk 11. And are now supported in FreePBX , But be careful even though the ‘hint’ says they can be numeric or names the just use the named variable.

namedcallgroup=office,home,1
namedpickupgroup=office,home

As above we have a namedcallgroup as 1 but this is not the same as callgroup 1

A named callgroup and pickupgroup can be set to a comma separated list of case sensitive name strings. The number of named groups is unlimited. The number of named groups you can specify at once is limited by the line length supported.

SYNTAX
namedcallgroup=[name[,name[,...]]]
namedpickupgroup=[name[,name[,...]]]
  • namedcallgroup – specifies which named pickup groups that this channel is a member.
  • namedpickupgroup – specifies which named pickup groups this channel can pickup.
Configuration Example
namedcallgroup=engineering,sales,netgroup,protgroup
namedpickupgroup=sales

Configuration should be supported in several channel drivers, including:

  • chan_dahdi.conf
  • misdn.conf
  • sip.conf
  • pjsip.conf

pjsip.conf uses snake case:

named_call_group=engineering,sales,netgroup,protgroup
named_pickup_group=sales

You can use named pickup groups in parallel with numeric pickup groups. For example, the named pickup group ‘4’ is not the same as the numeric pickup group ‘4’.

Numeric call pickup groups

(obsolete use named groups)

A numeric callgroup and pickupgroup can be set to a comma separated list of ranges (e.g., 1-4) or numbers that can have a value of 0 to 63. There can be a maximum of 64 numeric groups. This is important to note as Freepbx does not sanity check what you put in there, So you can put 70 in the Gui and it will show 70 but do a sip show peer or a pjsip show endpoint and you will see its not set.

Categories
Knowledge Base

Checking registered SIP peers

We got a call recently from a customer who uses skype trunks for some international incoming numbers. We have found that these lose registration on a regular basis, We could let them be set try and register indefinitely but this can have performance effects on the server.

This is not our only customer with this problem, We have also of late noticed a similar problem with Voip-unlimited where registration times out every day at 8pm, we have noted this on ours and other dealers servers.

So what to do?

Well we have written a quick script to check registration details and reload if not correct.

#!/bin/bash

HITS=`asterisk -r -x “sip show registry” | grep Registered |grep $1| wc -l`

if [ $HITS != $2 ]; then

echo “Incorrect number of $1 trunks are registered, we wanted $2  ”

echo “We will now reload asterisk ”

asterisk -r -x “reload chan_sip.so”

exit 0

else

echo “$HITS $1 trunks are registered, we wanted $2  ALL OK”

exit 0

fi

 

the syntax for running the script is

 

script_name.sh hostname 2

with hostname being part of the string you get when doing a sip show registry.

and the number is the number of registered trunks you expect with that name.

if all is well:”2 sipgate trunks are registered, we wanted 2 :-) ALL OK”and nothing happens

if not correct number then:

“Incorrect number of sipgate trunks are registered, we wanted 2

“We will now reload asterisk”

we only relaod the sip channel driver as this is enough to sort the problem normally

Categories
Technical

Reloading Aastra firmware

The Aastra Web Recovery Mode allows you to force an Aastra handset to re-download configuration and firmware files.  The phone enters web recovery mode if either of the following conditions occurs:

  • An incomplete download of the *.st binary file
  • A set being power cycled many times

When an Aastra Handset is in web recovery mode, the power will appear to come on the phone, but the LCD screen will remain completely blank, and you will be unable to use the phone.

If your Handset resembles  these symptoms, please do the following:

  1. Disconnect the Aastra phone from power or network in case of poe handsets
  2. Reconnect the power or the poe cable and  press and hold 1 + #
  3. Hold the keys listed in Step 2 until the phone displays an IP address on the LCD screen
  4. Using a computer on the same subnet as the phone, browse to the IP address displayed on the LCD screen
  5. You should see two fields on this page:
    1. Filename:
    2. TFTP Server IP:
  6. In the Filename field, enter the name of the phone’s firmware file.  Aastra firmware files are named <Model#>.st.  
    1. Firmware files are:
      51i.st
      53i.st
      55i.st
      57i.st
    2. New firmware can be obtained from www.aastratelecom.com website
  7. In the TFTP Server IP field, enter the IP address of your TFTP server or IPPBX.
  8. Click Download Firmware
  9. The phone make take a while to contact the server and load firmware and configuration files.
  10. The set will now restart.
Categories
Knowledge Base Technical

Nagios plugin for reading the Asterisk Database

This is a simple plugin that is based on one by Jason Rivers We have changed it now to read the ASTDB (Asterisk internal Database and then based on ok and Critical keys it will report OK or Critical staus reports to Nagios.

This was written for reporting if an Elastix system is in Day or Night mode.

You can define the Database Family, Key, Critical value and OK value. This means you can cutomise it to what ever you need to report.

 

The Code is below, make you may need to change /usr/bin/nc for what ever you use for netcat.

any issues email us, but dont forget this is given for free not supported for free.

#!/bin/bash
#
# Program : check_asterisk_ami
# :
# Author : Original code by Jason Rivers < jason@jasonrivers.co.uk >
# : Modified by Cyber-cottage.co.uk for checking the asterisk Database
# :
# Purpose : Nagios plugin to return Information from an Asterisk host using AMI
# :
# Parameters : --help
# : --version
# :
# Returns : Standard Nagios status_* codes as defined in utils.sh
# :
# Licence : GPL
#
# Notes : See --help for details
#============:==============================================================
PROGNAME=`basename $0`
PROGPATH=`echo $0 | /bin/sed -e 's,[\/][^\/][^\/]*$,,'`
REVISION=`echo '$Revision: 1.1.0.6 $' | sed -e 's/[^0-9.]//g'`
. $PROGPATH/utils.sh
print_usage() {
echo "Usage: $PROGNAME [-H hostname] [-u username] [-p password] [-P port] [-k DBkey] [-c critical] [-o ok] [-f family]"
echo " -H Hostname"
echo " -u AMI Username"
echo " -p AMI Password"
echo " -P (optional) AMI PORT"
echo " -k Database key"
echo " -f Database family"
echo " -c Critical Key"
echo " -o OK KEY"
echo ""
echo "SupportedCommands:"
echo " Most DB familiys that toggle such as DayNight in elastix"
echo "Usage: $PROGNAME --help"
echo "Usage: $PROGNAME --version"
}
print_help() {
print_revision $PROGNAME $REVISION
echo ""
echo "Nagios Plugin to check Asterisk ASTDB using AMI"
echo ""
print_usage
echo ""
echo "Asterisk Call Status Check. orignal version by © Jason Rivers 2011 changes to do ASTDB by cyber-cottage.co.uk"
echo ""
exit 0
# support
}
# If we have arguments, process them.
#
exitstatus=$STATE_WARNING #default
while test -n "$1"; do
case "$1" in
--help)
print_help
exit $STATE_OK
;;
-h)
print_help
exit $STATE_OK
;;
--version)
print_revision $PROGNAME $REVISION
exit $STATE_OK
;;
-V)
print_revision $PROGNAME $REVISION
exit $STATE_OK
;;
-H)
REMOTEHOST=$2;
shift;
;;
-P) AMIPORT=$2;
shift;
;;
-u) AMIUSER=$2;
shift;
;;
-p) AMIPASS=$2;
shift;
;;
-c)
CRITICALNAME=$2
shift;
;;
-o)
OKNAME=$2
shift;
;;
-k)
DBKEY=$2;
shift;
;;
-f)
FAMIL=$2;
shift;
;;
*)
echo "Unknown argument: $1"
print_usage
exit $STATE_UNKNOWN
;;
esac
shift
done
if [ "${AMIPORT}" = "" ]; then
AMIPORT="5038"
fi
if [ "${FAMIL}" = "" ]; then
##WARNING
echo="CRITICAL: Unknown KEY"
print_help
exit=$STATE_CRITICAL
else
## Checking Astdb
CHANNELS=`/bin/echo -e "Action: login Username: ${AMIUSER} Secret: ${AMIPASS} Events: off Action: DBGet Family: ${FAMIL} Key: ${DBKEY} Action: Logoff " | /usr/bin/nc $REMOTEHOST ${AMIPORT} | awk '/Val/ {print $2}'|tr -d " "`
if [ "$CHANNELS" = "" ]; then
echo "UNKNOWN: Unable to get ASTDB status"
exit $STATUS_UNKNOWN
fi
if [ "$CHANNELS" = "${OKNAME}" ]; then
exitstatus=$STATU_OK
MSG="OK: ${DBKEY} Asterisk Emergency message not active"
elif [ "$CHANNELS" = "" ]; then
exitstatus=$STATU_WARNING
MSG="WARNING: Asterisk Unknown status"
elif [ "$CHANNELS" = "$CRITICALNAME" ]; then
exitstatus=$STATU_CRITICAL
MSG="CRITICAL: ${DBKEY} Asterisk Emergency message active"
fi
fi
echo $MSG
exit $exitstatus
Categories
Support Technical

Completing your number porting form

This guide is designed to help you understand the porting process and provide us with the correct information to submit your port request to your current provider for their approval. The port forms can be downloaded from the Keep Your Number section of the Gradwell website, there is also a link to the port forms at the bottom of this article.

We will try to submit the most accurate port request, based on the information that you provide in the port form; however, you may find it useful to spend some time confirming the details with your current provider, prior to submitting the forms. This should reduce the likelihood of us receiving a rejection from the current provider, and a subsequent increase in the porting fees.

When you have completed the form please email it to porting@gradwell.com, along with a copy of a recent bill (issued within 3 months) from your current service provider, as proof of ownership of the numbers. If you do not receive paper bills then you can also provide a screenshot of an online account or an email/letter from your current provider, confirming the services that you have with them.

Once we have submitted your request we will send you a confirmation email. We will also keep you updated on our progress and inform you as soon as the port is accepted and a port date confirmed. Prior to the port date, we will set up the number in your account and add it to the package of your choice, to ensure a seamless transfer once the number ports.

1: CUSTOMER INFORMATION

Contact Name: The name of person that Gradwell will update as part of the porting process

Company Name (If applicable): Company name showing on your Gradwell account

Contact Email Address: Email address that Gradwell should send updates to

Contact Phone Number: Phone number that should be used by Gradwell, when contacting you to update you on your port

Gradwell Customer ID: This should be the account number of the Gradwell account, that the numbers will be added to prior to the port completing

2: INSTALLATION ADDRESS

All port requests must include an associated address to be submitted to your losing provider:

  • If you currently own a physical circuit, then the address we require is normally regarded as the installation address (where the physical circuit is installed)
  • If your calls are forwarded remotely from an exchange, then we may require the address and postcode of the exchange in question
  • If your numbers are currently on another VoIP platform, then your current provider may have associated a different address against the numbers, in lieu of a permanent installation address

If you are unsure of the information we require then we strongly suggest that you contact your current provider to check, as the address submitted must match what your current provider has on record or the order will be rejected and you will be charged a resubmission fee.

3: ABOUT YOUR CURRENT TELEPHONE SERVICE

Current Service Provider: The provider who is currently billing you for the number(s) to be ported

Account Number: The account number of your account with the current provider of the number(s) to be ported

Account Name: The account name of your account with the current provider of the number(s) to be ported

Service Type (e.g. Single Line, PBX system, Feature Line, DDI, ISDN line etc.): If you are unsure of the service type provided, please confirm this with your current provider

Single Analogue installation or Multi-line installation: If you are unsure whether your circuit is a single or a multi-line installation, this will need to be confirmed with your current provider

Are there any products associated with your current service? (e.g. Broadband, Redcare, etc.):

  • If you are porting all numbers from a physical circuit, then the physical line is likely to be ceased on completion of the port.
  • If you currently have a Redcare service from BT, or any other similar security alarm product, then this must be moved to another available line, prior to us submitting the port request to your losing provider. If it is not moved or ceased, then the order will be rejected and you will be charged a resubmission fee
  • If you currently have a broadband service on the line, then this will be ceased once the number port has completed. Gradwell will not be held liable for any services ceased on completion of the port.

If you are unsure whether there are any existing products, then we strongly suggest that you check this with your current provider.

4: MAIN BILLING NUMBER

What is the Main Billing Number? This is the main number listed on your account with your current provider

Do you want this number to port to Gradwell, be ceased or remain live with your current provider? A Main Billing Number can be ported, ceased or remain live with the current provider. It is important to remember that, if you are porting the main billing number away from your current provider, then any numbers associated to it may cease to work. If you are unsure whether you have any associated numbers, then this can be checked with your current provider.

5: ASSOCIATED NUMBERS

Are there any additional numbers associated to the Main Billing Number above? If you have a multi-line installation, then you may find that you also have other single numbers or DDI ranges associated to the main billing number. If this is the case then we must be informed of all associated numbers and confirmation provided as to whether they are to port, cease or remain live.

It is important to remember that, if you are porting the main billing number, then associated numbers may cease to work on completion of the port. In this instance, you may need to either port or cease these numbers. If you are unsure then please check this with your current provider.

6: PORTING DETAILS

Requested Port Date: If no date is listed then we will request the earliest possible port date, based on Ofcom mandated porting lead-times. The port will take place from Monday to Friday (excluding Bank Holidays), we do not complete any ports on a weekend. Occasionally the port date can change, however this will be communicated to you as soon as we have been notified by the losing provider.

Package Required for Ported Numbers: If you are unsure of the package that you require, please contact our Sales Team, who will be able to determine the correct package for your needs. The requested package will be set up in advance of the port taking place, to ensure a seamless transfer of your number.

Maintain Existing Directory Listing?: If your numbers are currently listed in the BT PhoneBook and you would like us to maintain this listing, please specify ‘Yes’ above. If your numbers are not listed, but you would like us to arrange for them to be included in the BT PhoneBook, please contact us on 01225 800 870 to discuss.

Onboarding Assistance Required?: At Gradwell, we offer additional support to help new customers familiarise themselves with our VoIP products and services. This includes assisting with the setup of softphones or hardware that has been purchased from Gradwell, and providing an overview of your new Gradwell control panel.

7: PORTING CHARGES

This section details the charges involved in porting your number(s) to Gradwell. By signing the declaration in section 8, you are agreeing to pay these charges.

8: DECLARATION

This section must be signed by the owner of the numbers to be ported, and by the person who will be paying for porting charges and the ongoing monthly recurring charges, once the number(s) have ported to Gradwell. If this is not the same person then both the current owner of the numbers, and the new owner of the numbers must sign the port form.

Categories
Calls and Lines Connectivity Installation Support Technical

Number Porting

When you switch to Gradwell, you can enjoy all the great benefits of our award-winning VoIP telephony service and keep your existing number. Becoming a Gradwell VoIP customer doesn’t mean losing your existing PSTN phone number. Our dedicated porting team can transfer your existing UK and international numbers within days of joining us to minimise any disruption to your business.

Cost: £20.00 one off fee to transfer your number, capped at £200 if you have a lot of numbers in the same sequence to move.

£20.00 each port rejection.

£10.00 for any date change/cancellation.

We are able to move your number from various telephony providers, including:

Full geographic porting:

  • Affiniti
  • BT
  • Cable & Wireless
  • Cable & Wireless (Energis)
  • Cable & Wireless (Thus)
  • Eurobell
  • Inclarity
  • Gamma
  • Magrathea
  • Opal (if Opal are not the range holder)
  • Spitfire
  • Viatel

Partial geographic porting agreements:

  • Colt
  • NTL
  • Telewest
  • Verizon

Full non-geographic (08) porting agreements:

  • Affiniti
  • Cable & Wireless (Energis)
  • Gamma
  • Magrathea
  • Opal
  • Viatel

We are always building on our porting agreements so if you have a number from a company not listed above, please let us know and we will see if it is possible to port.

To keep your number, you will need:

  • A completed porting form (If you require help completing the number porting form, please visit the How To article, which will guide you through the process.)
  • A copy of latest phone bill for the number(s) you want to keep, showing the main billing number and your company name.
  • To fax both of those documents to us on scan and email to support@cyber-cottage.co.uk
  • If you need to port 10 or more numbers you will need to complete the Capacity Planning Form, which is located on the second part of the Number Porting Form.

What are the timescales?

(Please note this is a guide and times are may be subject to delays)

  • Single line request – five working days
  • Multi line request (less than ten lines/numbers) – seven working days
  • Multi line request (more than ten lines/numbers) – seventeen working days
  • Simple DDI request – seventeen working days
  • Complex DDI request – twenty two working days
  • Single line SUP* request – seven working days
  • Multi line SUP* request (less than ten lines/numbers) – ten working days
  • Multi line SUP* request (more than ten lines/numbers) – twenty working days
  • Simple DDI SUP* request – twenty working days
  • Complex DDI SUP* request – twenty five working days
(A SUP request will be raised where the range holder and losing current provider are different)

Please note, any resubmissions or changes will be charged at £20 (excluding VAT) and will be subject to a new lead time. You will also be charged this amount for any numbers you port, but then do not use or cease. It may therefore be beneficial to cease the numbers with the current providers before putting through the port request to minimise cost.

For more information regarding porting your number, please visit the FAQs.

Categories
Knowledge Base Technical

Installing Webdav on Centos with untrusted ssl certificates

Webdav (Web-based Distributed Authoring and Versioning) is a set of methods based on the Hypertext Transfer Protocol (HTTP) that facilitates collaboration between users in editing and managing documents and files stored on World Wide Web servers.This can be useful to allow backing up of data between servers. In Linux there is a command line client called cadaver that in theory allows you to script its use.

This isnt as staight forward as it could have been and we spent half a day and a lot of Googling to get to the bottom of common problems, Such as auto loging in, Untrusted ssl certificates and scripting.

So here is a simple run down on what you have to do to script with webdav and cadaver

Install cadaver

yum install cadaver

once installed you will need to install the certificate for the untrusted ssl site

for example
wget http://website/untrusted.server.com.cer
now convert this to a .pem file
openssl x509 -inform der -in untrusted.server.com.cer -out untrusted.server.com.pem

now add this to your cert.pem file.

in centos this seems to be in the  /usr/share/ssl/ directory
cat untrusted.server.com.pem >> /usr/share/ssl/cert.pem

you now need to edit the ./netrc file with the server and logon details
vi ~/.netrc

machine untrusted.server.com
login   user
password        secret

save the file.

now when you connect you wont be promted for accepting the certificate or a username and password

cadaver -et  https://untrusted.server.com
dav:/>

Ok now we want to script the actions.
This is as simple as creating a script file.

for example

vi ~/.cadavscript
ls
pwd
quit

will give the following output

cadaver -et  https://untrusted.server.com/ < ~/.cadavscript
dav:/> Listing collection `/’: succeeded.
test.txt                              35  Aug 19 15:04
dav:/> Current collection is `https://untrusted.server.com/’.
dav:/> Connection to `untrusted.server.com’ closed.

So create a script to do what you want and your done

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