From the 1st September 2015 Yealink units purchased from us as an official Yealink (UK) reseller, will carry a Lifetime Warranty.
This Life Time Warranty will commence from the date of purchase; until 12 months after the official end of life date of that product.
This Lifetime Warranty excludes the following products:
– T18P, T20P, T20PN, T22P, T22PN, T26P, T26PN, T28P, T28PN, T32G, T32GN, T38G, T38GN, EXP38, EXP39, YHS32 and all VP series units.
– All Spare Parts including but not limited to : Network Cable (CAT5), Base Stand, Handset, Handset Curly Cord, Power Supply Unit (PSU)
The products listed above will be covered by the current 12 months from date of purchase or 24 months from date of manufacture, whichever is the longer.
Any extended warranty purchased prior to 1st September 2015 will continue to be honoured until the end of the purchased warranty period.
This warranty covers manufacturing faults and component defects. It does not cover wear and tear, physical damage, or problems associated with the network or platform it is connected to.
Please contact the distributor or ITSP the unit was purchased from for further details.
In the Event of a Failure
In the unlikely event that you have a faulty unit, you should contact us first.
We may require some additional information or additional testing and procedures to be carried out to establish the unit is faulty and one which they have supplied.
If the unit is deemed to be faulty we will request the unit is returned to us at your cost and subject to further tests, confirming the unit is faulty and that the warranty seal hasn’t been broken, a replacement unit will be supplied at no cost.
If the model returned under the Lifetime Warranty is no longer available we will supply an equivalent or better model.
This warranty is in addition to any statutory rights.
We recently had a very puzzling issue with a customer who we supplied some T23 Yealink handsets. When making outgoing calls over Gamma sip trunks on their Elastix server we were getting one way audio, This was not an issue with their existing Snom handsets or a problem for internal or incoming calls over the same trunks. It also wasn’t an issue when using iax2 trunks.
It seems that there is some interoperability issue when using sip trunks and these handsets. and seems to be a little known issue as only affects a few operators.
It seems to addressed in 44.80.0.20 version software that isn’t on the Yealink UK site yet but is available here and should be loaded on all T23 handsets as they are being delivered as 44.80.0.5 firmware at the moment.
In FreePBX users can listen to wav file recordings via the “Call Recordings” tab, This uses a field in the mysql cdr table to say where that recording is and what its called, They are now stored in year/month/day directory structure under /var/spool/asterisk/monitor so if the end user wants the recordings in mp3 format as many do its not just a case of converting them its also a case of updating the database.
Luckily this is fairly straight forward, its just a case of doing a quick query and then converting the file and the updating the database. First you have to install lame, This can be done simply with yum then write a script.
In FreePBX advanced settings, you need to enable “Display” and “Override” readonly settings and then add
The script I use is simple with a bit of basic logging.
#!/bin/bash
. postrecconfig.sh
date >> /var/log/asterisk/mp3.log
pcmwav=$(mysql -u$user -p$secret -s -N -D asteriskcdrdb<<<"select recordingfile from cdr where linkedid LIKE '$1' AND disposition = 'ANSWERED' ORDER by calldate DESC LIMIT 1");
mp3="$(echo $pcmwav | sed s/".wav"/".mp3"/)"
nice lame -b 16 -m m -q 9-resample "$path$pcmwav" "$path$mp3" >> /var/log/asterisk/mp3.log
touch -r "$path$pcmwav" "$path$mp3" >> /var/log/asterisk/mp3.log
mysql -u$user -p$secret -s -N -D asteriskcdrdb<<<"UPDATE cdr SET recordingfile='$mp3' WHERE recordingfile = '$pcmwav'" >> /var/log/asterisk/mp3.log
echo $pcmwav >> /var/log/asterisk/mp3.log
echo "--------||-------" >> /var/log/asterisk/mp3.log
date >> /var/log/asterisk/mp3.log
echo "Done" >> /var/log/asterisk/mp3.log
echo "--------||-------" >> /var/log/asterisk/mp3.log
exit 1
The postrecconfig.sh file looks like
user=freepbxuser
secret=secret
receptemail=info@youremailaddress.com
file_age=35
dy=$(date '+%Y')
dm=$(date '+%m')
dd=$(date '+%d')
path=/var/spool/asterisk/monitor/$dy/$dm/$dd/
As can be seen it steps through entry by entry converting and updating the DB, This example is cron'd to run hourly but does not delete the original wav file, this would be done in a separate script run weekly to remove old files. The reason to keep them is so that a backup of the original is held for a period in case of errors.
Dynamic features are very useful for allowing users access to custom features during calls. These can be loaded individually via the dialplan, but in freepbx based solutions this will mean a bit of hacking of the dialplan using overides and making sure all still works afterwards, or as a global varible.
The easiest way is to load them as a global as is done with apprecord, But if you want to add lots of features then you will have to use a Application Map group. This is done by editing the features_applicationmap_custom.conf file so it looks like below for example, at the top are your application maps then your group
The SIP2SIM service is a very simple concept which provides you with control of your mobile communications. It is ideal for an office of any size and even for more technical home users.
The service consists of a SIM card, which you put in a mobile phone and it makes that phone appear as if it is a SIP extension (e.g. SIP phone) on a phone system of your choice.
An extension on your office phone system
The basic service allows you to specify, on our control pages, the server name, login and password for a SIP server. This could be your office phone system whether an asterisk box, or a FireBrick or whatever. As long as it handles normal UDP SIP with G.711 a-law audio then we will register as a phone and allow calls both ways.
This means your phone can simply be an office extension, like any other.
Call office extensions using short extension numbers from your mobile phone.
Office policies on callable numbers, such as premium rate, enforced like any other extension.
Office voicemail system working, just like any other extension.
Office call logging, just like any other extension.
Office call recording, just like any other extension.
Use in hunt groups, just like any other extension.
Even use features like call steal to transfer calls if you want, or in-band DTMF to control call transfer and related features.
Manage costs
The costs are very simple for using the SIM in the UK on O2. Higher costs apply for roaming SIMs, even roaming to other UK networks.
£5+VAT to purchase the SIM card.
£2+VAT pcm once activated (activates on first use).
2p+VAT per minute for calls either way.
2p+VAT per text either way.
2p+VAT per MB for data usage either way.
No minimum term – just monthly rolling contract.
Calls are charged per second, and data per kilobyte. Charges apply to the SIP2SIM service separately from any charges you may have for text or voice calls using your VoIP/SMS provider.
In some cases a SIM will be set up with a free trial which allows some usage without assigning to an account or setting up an account. For these trial SIMs, once assigned to an account an activation fee of 1p+VAT is charged and usage charges commence as normal.
Telephone numbers
Just like a SIP handset, the service does not come with any sort of telephone number.
You can, if you wish, have the phone register on a VoIP provider’s service. This would mean you get calls to a number operated by the VoIP provider, and can make calls from that number, just like any other SIP phone. If registered with your own telephone system, it would have internal extension numbering, and even direct dial in numbering as you have chosen to configure on your phone system.
There is no question of porting numbers to or from the service, it has no numbers.
We can, of course, provide telephone numbers as part of our VoIP service if you wish, and we can even pre-configure these to connect with your SIM as part of the order process.
International roaming
The SIMs are available with world wide roaming. Costs are higher when roaming, obviously. The following are charges for use within EU. See full roaming price list for more details.
£5+VAT to purchase the SIM card.
£2+VAT pcm once activated (activates on first use).
10p+VAT per minute for calls either way.
5p+VAT per text either way.
10p+VAT per MB for data usage either way.
No minimum term – just monthly rolling contract.
In the UK and rest of EU, calls are charged per second, and data per kilobyte. Charges apply to the SIP2SIM service separately from any charges you may have for text or voice calls using your VoIP/SMS provider.
UK roaming for the best coverage of the UK
The SIMs can even roam in the UK. This means the SIM can change its identity automatically – being an O2 SIM when O2 is available (at the lowest costs), but switching to a foreign identity (Dutch Vodafone) to roam on to other UK networks. Whist costs are then higher this means you stay in touch even when there is no O2 signal.
When the SIM is using the O2 profile and on O2 in the UK, the UK prices apply. When using the EU profile on any UK network (including O2) or elsewhere in the EU, the EU roaming prices (as above) apply. If on EU profile on O2, it will normally switch back to O2 profile automatically within a few minutes. Currently the SIM will not roam to Three in the UK.
Text
Texts are operated separately. Our text interface using HTTP can send texts to the phone, and texts from the phone can be posted to an HTTP gateway of your choice. If you have an A&A VoIP telephone number then texts can be associated with that number directly (not all of our numbers ranges can handle inbound texts).
Mobile Data
Data currently allows simple NAT, unfiltered, Internet access. We hope to offer data via A&A in the future.
Third Party SIP services
The service involves entering SIP registration details in to our control pages. Where these are the details of your own SIP server such as an office phone system, you can make the decision as to whether or not you trust us with those SIP details in order to provide the SIP2SIM service. We will, of course, use all reasonable skill and care to ensure the details remain confidential and are not disclosed.
We have, however, designed the service so that it can work with a wide variety of third party SIP services, not just in the UK but in various countries. There are a lot of unusual systems out there and we continue to work to ensure that such services operate with SIP2SIM. However, using third party SIP details may well be in breach of your terms with the third party SIP provider as it means giving us your SIP details. It is up to you to check the terms and we would not suggest anyone breaks a contract they have. You may find that the provider is happy to trust us, especially if they do any other work with us, so it is worth asking. We are also happy to discuss contracts with other providers for the SIP2SIM service and we may be able to provide them with branded SIMs to sell to their customers.
Unfortunately we cannot guarantee that the service will always work with all other providers, and whilst we aim to resolve any technical issues (with reference to the standards), if a provider simply will not deal with our service and blocks us there is little we can do.
The Sip2Sim service is provided by Andrews and Arnold Ltd and can be purchased direct from them or via ourselves where we will assist with setting up the service for you.
We run many scripts on customer servers to email cdrs, backups etc, one problem with some mail servers is the mail gets rejected as it comes from root@elastixserver.yourdomain.com by default to fix this is simple and only takes a few lines.
Postfix MTA offers smtp_generic_maps parameter. You can specify lookup tables that replace local mail addresses by valid Internet addresses when mail leaves the machine via SMTP.
Open your main.cf file
# vi /etc/postfix/main.cf
Append following parameter
smtp_generic_maps = hash:/etc/postfix/generic
Save and close the file. Open /etc/postfix/generic file:
# vi /etc/postfix/generic
Make sure root@elastixserver.yourdomain.com change to elastixserver@yourdomain.com add :
Save and close the file. Create or update generic postfix table:
# postmap /etc/postfix/generic
Restart postfix:
# /etc/init.d/postfix restart
When mail is sent to a remote host via SMTP this replaces root@elastixserver.yourdomain.com by elastixserver@yourdomain.com mail address. You can use this trick to replace address with your ISP address if you are connected via local SMTP.
Since Version 1.8 in Asterisk we have seen some issues with DID calls from some suppliers.
The tell tail sign is that even though you have an inbound route that matches the DID it will still say in the verbose screen that nothing matched it in the inbound context, For example:-
Call from 'USERNAME' (XXX.XX.XXX.XX:5060) to extension '01234123412' rejected because extension not found in context 'from‐trunk'
and if you do “dialplan show 01234123412@from-trunk” sure enough there is one.
After much searching and experimentation below is a working freepbx config that has been tested with 1.8 and 11 and proves to be working with the suppliers above.
QueueMetrics, The Advanced Call Center Software Solution Suite. Measure your targets, conversion rates and agent activities. Create accurate reports and statistics. Set security and privacy on individual queues. Support virtual and multi-tenant production environments.
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QueueMetrics Features:
Live administrator and supervisor call center status panel.
Area code breakdowns inclusive of calling and waiting time.
Agent billable and payable time with total sales, contacts and conversion statistics.
Live agent page with queue statistics and agent controls.
Total unanswered calls with disconnection time and position.
Complete call distribution statistic, including sales and contacts, by week, day or hour.
Administrator message broadcasting and SMS functionality.
Full agent availability with session and pauses details and history.
Inbound ACD call attempts with metrics available by operator, terminal and queue.
Detailed call information including the Asterisk Call ID and recorded call.
Total of answered calls including call length and waiting time metrics.
Inclusive SLA of answered and unanswered calls and disconnection causes.
Extensive Quality Assessment module.
Send automated nightly PDF/XLS exports by e-mail.
Hundreds of metrics computed.
Operations Managers can:
See accurate reports of all call center activities.
Run reports by single and by user-created queue groups.
Measure agents activities, business targets and conversion rates.
Fully configure security and privacy, queue-by-queue.
Team Leaders can:
Create real time call and agent reporting.
See agent status and realtime activities.
Remotely listen to live calls as they are handled.
Watch agent screens through a VNC client.
Agents can:
See the calls they’re handling and integrate with external CRM.
Pass data gathered from IVR menus or CallerID.
Set call status codes for all inbound and outbound traffic.
Logon, logoff, go on pause and set pause reason codes.
IT Managers can:
Support single-server or Asterisk® clusters.
Support database and flat-file storage.
Tune Asterisk® interaction to minimize the load on the Asterisk® server.
Avoid patching or changing an existing Asterisk® installation.
To download a product feature sheet click here or call us for a quote.
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QueueMetrics, The Advanced Call Center Software Solution Suite. Measure your targets, conversion rates and agent activities. Create accurate reports and statistics. Set security and privacy on individual queues. Support virtual and multi-tenant production environments.