IPecs 1000i range phone provisioning

 I have found that from time to time phones connected to IPecs cloud systems will fail to connect. They sit there at a screen saying No Ack and giving you a range of options.

Usually forcing global provisioning again gets them working.

On the screen it boots to press *  for the [SET] option

It asks for a password – enter 147*

Select Option3 – Default

It asks, Default all profiles, are you sure?

Select YES

Press the BACK button

Reboot Now = YES

Shortly into the reboot the words “Global Provisioning” will display . Hit the Yes button

Hopefully this will help get someone back up and running.

LetsEncrypt failure on Draytek Routers

 We like to use Draytek Routers on our installs as they are easy to configure and tend to be reliable.

As with all things these days https secure connections have been enforced by Chrome and the browser world. This means the routers have to be accessed on https.  This then complicates things as you then need a valid ssl certificate.

Draytek have built in a LetsEncrypt client in their routers, so we use DynDDNS and LetsEncrypt in order to give the router a certificate.  To obscure things further we changed the ssl port away from 443.

All this worked fine until LetsEncrypt changed things. Ou certificates would not renew. It used to be that LetsEncrypt renewd on port 80, but to get things working again we had to change the secure  port back to 443. Suddenly we could renew the cert.

Keep that in mind. To use a Draytek with LetsEncrypt you now need ssl on port 443!

John Rogers

www.oneszeros.biz

FreePBX paging timeout

 We had an issue this week with paging on FreePBX. The customer had a user who made a page to a group of handsets but did not hangup his extension properly. This meant that all the other extensions were locked busy for quite some time. 

The customer asked if we could set a maximum page length. We had a look and the GUI doesn't allow for it. That meant we would need to find a way to modify the config files.

Any config done in the GUI is written into etc/asterisk/extensions_additional.conf. Any config FreePBX writes into here can be overwritten in etc/asterisk/extensions_override_freepbx.conf.

I used WinSCP to get into the FreePBX file system and opened extensions _additional.conf. I dod a search for the page group number which was 2800. This took me to a context called [ext-paging] and below it [app-pagegroups]. I copied the two sections and pasted them into notepad so that I had got an original copy of the config.

Then I used WinSCP to open extensions_override_freepbx.conf and I pasted in the [app-pagegroups] section. It looked like this

[app-pagegroups]
include => app-pagegroups-custom
exten => 2800,1,Macro(user-callerid,)
exten => 2800,n,Set(_PAGEGROUP=2800)
exten => 2800,n(busy-check),GotoIf($[${TRYLOCK(apppagegroups2800)}]?:busy)
exten => 2800,n(devstate),Set(DEVICE_STATE(Custom:PAGE2800)=INUSE)
exten => 2800,n,Gosub(app-paging,ssetup,1())
exten => 2800,n,Set(PAGEMODE=PAGE)
exten => 2800,n,Set(PAGE_MEMBERS=1050-1499)
exten => 2800,n,Set(PAGE_CONF_OPTS=)
exten => 2800,n,Set(ANNOUNCEMENT=beep)
exten => 2800,n(agi),AGI(page.agi)
exten => 2800,n,Set(CONFBRIDGE(user,template)=page_user_duplex)
exten => 2800,n,Set(CONFBRIDGE(user,admin)=yes)
exten => 2800,n,Set(CONFBRIDGE(user,marked)=yes)
exten => 2800,n,Answer
exten => 2800,n(page),ConfBridge(${PAGE_CONF},,,admin_menu)
exten => 2800,n,Hangup
exten => 2800,n(busy),Set(PAGE${PAGEGROUP}BUSY=TRUE)
exten => 2800,n(play-busy),Busy(3)
exten => 2800,n(busy-hang),Goto(app-pagegroups,h,1)
exten => h,1,ExecIf($[${ISNULL(${PAGE${PAGEGROUP}BUSY})}]?Set(DEVICE_STATE(Custom:PAGE${PAGEGROUP})=NOT_INUSE))
;--== end of [app-pagegroups] ==--;

Then I just needed to add a timeout. About line 7 where all the Set statements start I added 

exten => 2800,n,Set(TIMEOUT(absolute)=30)

I then saved the file and issued a fwconsole reload to the terminal of FreePBX to get the config read in.

Now a page will cause the channel to be opened, but if it's not hung up then it will finish and hang up at 30 seconds.

The problem with this approach is if  I want to change the group I can't do it in the GUI anymore. I have noted that in the description of the page group, but if anyone can explain a better approach then I would be happy to hear it.

John Rogers

Ones and Zeros

www.oneszeros.biz

Mitel 5312, 5320, 5330 and 5340 BLF in SIP mode

 A few years ago I posted a how to guide to getting Mitel dual mode phones into SIP mode and registering them on your SIP server.  While my configuration on that particular job used the programable keys on the phone, there was no call for Busy Lamps.

Last week I was asked to use these phones and get the BLF working.  I scoured the Internet and having patched together a couple of resources managed to get them to work.

On these phones you have the <User> tag section to define your user and also the <pkDescription> section to define the keys. Just as your phone wont work for you without both the user and the key setup, so the busy lamps wont work without a modification in both of those sections.

In the <User> section you need to define the users you want to subscribe to BLF updates for. User the BLFGroup="" to do this. Fill the quotes with a comma separated list of extensions to monitor . Then in the <pkDescription> section you define a key as Fea="27" 

The below file is the MN_userID.cfg file in my TFTP server.  Note how I use the BLFGroup= to setup a subscription for 1050, then in the key section, key 27 (third up from bottom) is the BLF key for 1050

<Parameter Model="5312">

<user_list>

<User ID="1001" DispName="John-Rogers" Pwd="MyPassword" AuthName="1001" Realm="" RegSvr="sipserver.net" RegPort="5060" ProxyScheme="2" ProxySvr="sipserver.net" ProxyPort="5060" ProxyScheme="2" VMSvr="sipserver.net" VMPort="5060" VMScheme="2" OutSvr="" OutPort="5060" OutCtr="0" Ring="1" Line="0" EventSvr="" EventPort="5060" EventScheme="2" BLFGroup="1050">

</User>

</user_list>

<pkDescription>

<Key Line="25" Fea="6" Des="Line 1" Addr="" Mode="1" UserID="271581001"></Key>

<Key Line="26" Fea="7" Des="Line 2" Addr="" Mode="1" UserID="271581001"></Key>

<Key Line="27" Fea="27" Des="1050" Addr="1050" Mode="1" UserID="1050"></Key>

</pkDescription>

<vlan_id>0</vlan_id>

<host_name>SIP13</host_name>

</Parameter>

Have a play, it worked for me.

John Rogers

www.oneszeros.biz

Killing stuck channels FreePBX Asterisk

 I keep finding stuck channels in asterisk / FreePBX conferences, that I then have to manually hangup.

Here is how you find the calls

core show channels

This will bring up a list if channels in use. Identify the one you need to hangup i.e.

PJSIP/itsptrunk-00000002

Then issue command

channel request hangup PJSIP/itsptrunk-00000002

The call will then drop and you have killed the call.

If you need any help with a telephony project you can reach me at www.oneszeros.biz

Asterisk & FreePBX Feature Codes

I have tended to find that documentation for Asterisk and FreePBX based phone systems can be rather spread about. For the generic phone user what codes should they use? Here is a helpful list.

Call Forward

*72 – Call Forward All Activate
*73 – Call Forward All Deactivate
*93 - Call Forward All Prompting Activate
*74 – Call Forward All Prompting Deactivate

*90 – Call Forward Busy Activate
*91 – Call Forward Busy Deactivate
*94 - Call Forward Busy Promping Activate
*92 – Call Forward Busy Prompting Deactivate

*52 – Call Forward No Answer/Unavailable Activate
*53 – Call Forward No Answer/Unavailable Deactivate
*740 – Call Forward Toggle

So how do you call forward to voicemail? This is one point that seems to escape all the guides. Calls can be sent direct to a users voicemail by putting a * in front of their extension number, so you can forward direct to voicemail by dialling *72 *ext. e.g. if my extension is 1002 I can dial *72 *1002 to do the forward.

Turning the prompting on and off turns on and off the voice responses that walk you through it and confirm your actions.

Call Waiting

*70 – Call Waiting – Activate
*71 – Call Waiting – Deactivate

*8 – Asterisk General Call Pickup (When you are part of a call group)
** – Directed Call Pickup (When you are part of a call group)
*2 – (When in a call) Asterisk Attended Transfer
## – (When in a call) Asterisk Blind Transfer
** – (When in a call) Asterisk Disconnect Code
*1 – (When in a call) Asterisk Toggle Call Recording

If you need any help please contact me via www.oneszeros.biz

Mitel MSL Based Upgrade Corruption

Occasionally when upgrading items using the "Blades" Panel on a Mitel machine, you may get inability to proceed due to checksum errors or the like.

Follow the process below to delete the corrupt files and get back to a fresh set of upgrade files.

Environment

MSL
Mitel Standard Linux
MBG
Mitel Border Gateway
MiCollab

Cause

A cached package that forms part of the upgrade is corrupt.

Resolution

Access the package location via the secure shell and delete the files.

SSH onto the MSL server as user root using something like PuTTY and issue the following commands:

cd /var/cache/e-smith/blades/packages/
rm *.* -f

Go back to the MSL web interface and access the ServiceLink > Blades page
Select the "Update List" button to refresh the list of blades and cached downloads

You can now proceed to cache the download again before attempting to upgrade the server again.

John Rogers

www.oneszeros.biz