Categories
Asterisk Support Elastix Support Knowledge Base Support Technical

One way audio with Yealink T23 and Gamma Sip trunks on Freepbx

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.

Categories
Elastix Support Knowledge Base Support

Converting recordings to MP3 in FreePBX and updating mysql CDR records

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

/usr/local/sbin/postrecord.sh ^{CDR(linkedid)} to “

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.

Hope this is of help to you and your users

Categories
Asterisk Support Elastix Support Knowledge Base Support

Multiple Dynamic features with Asterisk Applicationmaps

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

testfeature => #9,callee,Playback,tt-monkeys 
calleehangup => #8,callee,Hangup()
callerhangup => #7,caller,Hangup()
[mymapgroup]
testfeature => #9
calleehangup => #8
callerhangup => #7
apprecord => *1

DO NOT FORGET to add the apprecord to your group.

You then need to edit the globals_custom.conf file and add a line like below

DYNAMIC_FEATURES => mymapgroup

Then reload asterisk and issue the command “features show”

Dynamic Feature           Default Current
---------------           ------- -------
callerhangup              no def  #7     
calleehangup              no def  #8     
testfeature               no def  #9     
apprecord                 no def  *1     
Feature Groups:
---------------
===> Group: mymapgroup
===> --> apprecord (*1,caller,Macro,one-touch-record)
===> --> callerhangup (#7)
===> --> calleehangup (#8)

and to check that they are loaded as a global variable do “dialplan show globals” and near or at the top you will see:-

 DYNAMIC_FEATURES=mymapgroup

And thats all there is to it.

Categories
Elastix Support Knowledge Base Technical

Setting the server domain in elastix correct for scripted email

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 :

root@elastixserver.yourdomain.com  elastixserver@yourdomain.com

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.

To set up gmail for delivery look at this

Categories
Asterisk Support Elastix Support Knowledge Base

Sip Config for Aretta CBeyond and Voiceflex with Asterisk

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.

OUTBOUND

[peername]
username=USERNAME
type=peer
trustrpid=yes
sendrpid=yes
secret=PASSWORD
qualify=no
outboundproxy=sip.hostname.com
nat=yes
insecure=very
host=sip.hostname.com
fromdomain=sip.hostname.com
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw

INBOUND

[username]
type=peer
host=sip.hostname.com
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw

;registration string
USERNAME:PASSWORD@peername/USERNAME
Categories
Asterisk Support Elastix Support Knowledge Base

IAX2 Cause code

Here is a table of the IAX2 to assist with debugging IAX2 call issues

More IAX2 information can be found here and the RFC is here


CSV
 download is here
Number Cause Reference
1 Unassigned/unallocated number [RFC5457]
2 No route to specified transit network [RFC5457]
3 No route to specified transit network [RFC5457]
4-5 Unassigned
6 Channel unacceptable [RFC5457]
7 Call awarded and delivered [RFC5457]
8-15 Unassigned
16 Normal call clearing [RFC5457]
17 User busy [RFC5457]
18 No user response [RFC5457]
19 No answer [RFC5457]
20 Unassigned
21 Call rejected [RFC5457]
22 Number changed [RFC5457]
23-26 Unassigned
27 Destination out of order [RFC5457]
28 Invalid number format/incomplete number [RFC5457]
29 Facility rejected [RFC5457]
30 Response to status enquiry [RFC5457]
31 Normal, unspecified [RFC5457]
32-33 Unassigned
34 No circuit/channel available [RFC5457]
35-37 Unassigned
38 Network out of order [RFC5457]
39-40 Unassigned
41 Temporary failure [RFC5457]
42 Switch congestion [RFC5457]
43 Access information discarded [RFC5457]
44 Requested channel not available [RFC5457]
45 Pre-empted (causes.h only) [RFC5457]
46 Unassigned
47 Resource unavailable, unspecified (Q.931 only) [RFC5457]
48-49 Unassigned
50 Facility not subscribed (causes.h only) [RFC5457]
51 Unassigned
52 Outgoing call barred (causes.h only) [RFC5457]
53 Unassigned
54 Incoming call barred (causes.h only) [RFC5457]
55-56 Unassigned
57 Bearer capability not authorized [RFC5457]
58 Bearer capability not available [RFC5457]
59-62 Unassigned
63 Service or option not available (Q.931 only) [RFC5457]
64 Unassigned
65 Bearer capability not implemented [RFC5457]
66 Channel type not implemented [RFC5457]
67-68 Unassigned
69 Facility not implemented [RFC5457]
70 Only restricted digital information bearer capability is available (Q.931 only) [RFC5457]
71-78 Unassigned
79 Service or option not available (Q.931 only) [RFC5457]
80 Unassigned
81 Invalid call reference [RFC5457]
82 Identified channel does not exist (Q.931 only) [RFC5457]
83 A suspended call exists, but this call identity does not (Q.931 only) [RFC5457]
84 Call identity in use (Q.931 only) [RFC5457]
85 No call suspended (Q.931 only) [RFC5457]
86 Call has been cleared (Q.931 only) [RFC5457]
87 Unassigned
88 Incompatible destination [RFC5457]
89-90 Unassigned
91 Invalid transit network selection (Q.931 only) [RFC5457]
92-94 Unassigned
95 Invalid message, unspecified [RFC5457]
96 Mandatory information element missing (Q.931 only) [RFC5457]
97 Message type nonexistent/not implemented [RFC5457]
98 Message not compatible with call state [RFC5457]
99 Information element nonexistent [RFC5457]
100 Invalid information element contents [RFC5457]
101 Message not compatible with call state [RFC5457]
102 Recovery on timer expiration [RFC5457]
103 Mandatory information element length error (causes.h only) [RFC5457]
104-110 Unassigned
111 Protocol error, unspecified [RFC5457]
112-126 Unassigned
127 Internetworking, unspecified [RFC5457]
128-255 Unassigned

 

Categories
QueueMetrics Support Software

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.

But above all Improve your business.

 

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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.

Categories
Elastix Support Security

SSLv3 Poodle and Elastix

Google has just disclosed SSL POODLE vulnerability which is a design flaw in SSLv3.  By default SSLv3 is enabled by default in Elastix and many other servers, Since it is a design flaw in the protocol itself and not an implementation bug, there will be no patches. Only way to mitigate this is to disable SSLv3 in your web server or application using SSL.

How to test for SSL POODLE vulnerability?

The following simple script will test, its a re-write of Redhats that would give a false negative if the script fails in anyway giving a false sense of security.

#!/bin/bash
chmod 755 /usr/share/doc/bash-3.2/scripts/timeout
ret=$(echo Q | /usr/share/doc/bash-3.2/scripts/timeout 5 openssl s_client -connect "127.0.0.1:${2-443}" -ssl3)
if echo "${ret}" | grep -q 'Protocol.*SSLv3'; then
 if echo "${ret}" | grep -q 'Cipher.*0000'; then
 echo "SSL 3.0 disabled"
 else
 echo "SSL 3.0 enabled"
 fi
else
 echo "SSL disabled or other error"
fi

The outputs will be similar to below on Elastix

[root@elastix24 ~]# ./sslv3.sh 
depth=0 /C=--/ST=SomeState/L=SomeCity/O=SomeOrganization/OU=SomeOrganizationalUnit/CN=localhost.localdomain/emailAddress=root@localhost.localdomain
verify error:num=18:self signed certificate
verify return:1
depth=0 /C=--/ST=SomeState/L=SomeCity/O=SomeOrganization/OU=SomeOrganizationalUnit/CN=localhost.localdomain/emailAddress=root@localhost.localdomain
verify error:num=10:certificate has expired
notAfter=Jun 15 18:30:20 2014 GMT
verify return:1
depth=0 /C=--/ST=SomeState/L=SomeCity/O=SomeOrganization/OU=SomeOrganizationalUnit/CN=localhost.localdomain/emailAddress=root@localhost.localdomain
notAfter=Jun 15 18:30:20 2014 GMT
verify return:1
DONE
SSL 3.0 enabled

As we can see its enabled.

Now edit the file  /etc/httpd/conf.d/ssl.conf

and change line 100 (in Elastix 2.4)

from SLProtocol all -SSLv2    to  SLProtocol all -SSLv2 -SSLv3

The restart the httpd service.

then test again and you should get

13033:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure:s3_pkt.c:1086:SSL alert number 40
13033:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake failure:s3_pkt.c:530:
SSL disabled or other error

If you want to read the background here is the relevant document

Click to access ssl-poodle.pdf

Categories
Asterisk Support Elastix Support Knowledge Base Security

Elastix 2.4 ARI vulnerability Patch

The recent vulnerability in the Asterisk and Freepbx ARI login.php file is not addressed in an update to ARI in the unembedded freepbx on Elastix 2.4.

This will mean that your systems will still be vulnerable.

We have produced a patch that you can apply to address this. The patch can be downloaded  from https://s3.amazonaws.com/filesandpatches/ari.patch and applied as detailed below.

logon to the server console

cd /var/www/html/recordings/includes
cp login.php /root/login.php.ari
wget https://s3.amazonaws.com/filesandpatches/ari.patch
patch < ari.patch 

Then to check either login to server ARI interface or 

cat login.php |grep json

and you should get the following output

$buf = json_decode($_COOKIE['ari_auth'],true);
$data = json_decode($crypt->decrypt($data,$ARI_CRYPT_PASSWORD),true);
$data = $crypt->encrypt(json_encode($data),$ARI_CRYPT_PASSWORD);
$buf = json_encode(array($data,$chksum));


also check to see if you have the file in the fw_ari directory.

ls -l /var/www/html/admin/modules/fw_ari/htdocs_ari/includes

if there is a login.php there then copy over the patched version.

cp /var/www/html/recordings/includes/login.php  /var/www/html/admin/modules/fw_ari/htdocs_ari/includes/login.php

After these actions check that the file ownership is still correct

if not 

chown asterisk:asterisk /var/www/html/recordings/includes/login.php 

This patch also applies to any older version of ARI out there.

also to be on the lookout for two suspicious files, named “c.sh” or “c2.pl” respectively. If you see these two files remove them immediately!

More details here. http://community.freepbx.org/t/critical-freepbx-rce-vulnerability-all-versions-cve-2014-7235/24536 or here http://support.freepbx.org/node/92822

 

 

 

Categories
Case Studies Knowledge Base QueueMetrics Support

QueueMetrics

We have recently installed and customised a Queuemetrics solution for a customer. Their key reason for choosing Queuemetrics was the ability to use dynamic agents without the need of major Elastix reprogramming and a clear and simple interface.

It was also decided that users needed to be able to log into the system from their handsets as well as from a web interface, as the customer is hoping to roll out an agent portal in the future.

To enable Hotdesk the  following setting has to be set similar to below

# The value is interval time (in seconds) used by the analyzer to look back searching HOTDESK verbs in the queue log
default.hotdesking=86400

This meant some additional dialplans to allow logging in & out and pausing.

These are similar to the dialplans that the web portals use except that they have prompts and they also have to store the extension and agent id in the asterisk database.

;added dialplan for queuemetrics
; Add Member - 422
; User is asked for their loging agent id
exten => _422XXXX,1,Answer
exten => _422XXXX,2,Read(AGENTID,agent-login,4,,1,6)
exten => _422XXXX,3,Gotoif($["${AGENTID}" = ""]?end)
exten => _422XXXX,4,GotoIf($[${LEN(${AGENTID})} != 4]?2)
exten => _422XXXX,5,set(DB(qmagent/${CALLERID(num)})=${AGENTID})
exten => _422XXXX,6,Macro(queuelog,${EPOCH},${UNIQUEID},NONE,Agent/${AGENTID},HOTDESK,SIP/${CALLERID(num)})
exten => _422XXXX,7,AddQueueMember(${EXTEN:3:4},SIP/${CALLERID(num)})
exten => _422XXXX,8,SayDigits(${AGENTID})
exten => _422XXXX,9,Playback(agent-loginok)
exten => _422XXXX,10(end),Hangup
; Remove Member - 423
exten => _423XXXX,1,Answer
exten => _423XXXX,2,set(DEL_AGENT=${DB_DELETE(qmagent/${CALLERID(num)})})
exten => _423XXXX,3,RemoveQueueMember(${EXTEN:3:4},SIP/${CALLERID(num)})
exten => _423XXXX,4,Playback(agent-loggedoff)
exten => _423XXXX,5,Hangup
; extension 32: agent pause with hotdesking (with pause code)
exten => _32XX,1,Answer
exten => _32XX,2,set(AGENTCODE=${DB(qmagent/${CALLERID(num)})})
exten => _32XX,3,NoOp( "QM: Pausing Agent/${AGENTCODE} at extension SIP/${CALLERID(num)} with pause reason '${EXTEN:2:2}' made by '${QM_LOGIN}' " )
exten => _32XX,4,PauseQueueMember(,SIP/${CALLERID(num)})
exten => _32XX,5,System( echo "${EPOCH}|${UNIQUEID}|NONE|Agent/${AGENTCODE}|PAUSEREASON|${EXTEN:2:2}" >> /var/log/asterisk/queue_log )
exten => _32XX,6,Playback(dictate/paused)
exten => _32XX,7,Hangup
; extension 33: agent unpause with hotdesking
exten => 33,1,Answer
exten => 33,2,NoOp( "QM: Unpausing Agent/${AGENTCODE} at extension SIP/${CALLERID(num)} made by '${QM_LOGIN}' " )
exten => 33,3,UnpauseQueueMember(,SIP/${CALLERID(num)})
exten => 33,4,Playback(dictate/pause)
exten => 33,5,Playback(removed)
exten => 33,6,Hangup

These need to be added to your extensions_custom.conf file in a context that’s included in the from-internal  context.

Also a change has to be made to the dialplans in the extensions_queuemetrics.conf to store and delete the database entry as well.

The system has proved to deliver what was expected and will shortly be expanded to track outbound calls and the addition of custom wallboards similar to what we recently produced for another customer.

If you would like to talk about adding QueueMetrics to your Asterisk system or are looking for a complete phone system and queuemetrics platform please contact us.