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Different Operator when you Dial 0 based on extension..

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I have one I haven't run into yet, I have a company that wants to have different assigned operators (if a desk phone dials a 0), depending on what extension they are calling from.

So say if your calling from extension 3xxx, then if you hit 0 get get sent to the operator at extension 3100, but if your calling from an extension in the 8xxx range, and you pick up your phone and dial 0, then you get sent to the operator at extension 8200.

Any easy way to do this in FreePBX/FPBX-Distro outside of a bunch of custom dialplan and contexts?


.call file to pipe audio to conference room at scheduled time

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Hi There -

I'm running latest freepbx 2.11.0.27 at the moment although I dare say what I'm asking for isnt version specific.

I have a number of users in a conference meeting at as soon as a mp3 stream goes live I need to pipe it into the conference room (using MeetMe for the time being, but no particular reason why I cant using ConfBridge if necessary)

I've been reading the asterisk 1.4-1.6 training book from http://the-asterisk-book.com/ but I'm missing something

Heres my context at the moment:

[custom_playstream]
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Playback(/var/lib/asterisk/sounds/en/what-time-it-is)
exten => s,n,MeetMe(8000,x,secretpassword)

And heres my test .call file I'm pumping into the spool output area:
Channel:local/101010
Callerid: 101010
MaxRetries: 5
RetryTime: 300
WaitTime: 45
Context: custom_playstream
Extension: s
Priority: 1

Extension 101010 is just a virtual extension and I think this might be one of my problems (dont think you can answer a virtual extension??)

I think my main problem is it seems to assume the call is originating from 101010 whereas I don't need it to originate from anywhere. Sorry if this is already covered, but I've been looking everywhere and google searched for items like auto dial out , connect call to conference etc

Multiple Parking Lots, Valet Parking

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FreePBX is great.

I wanted to set up multiple parking lots and direct calls into a particular lot, but I could not find how to do this in FreePBX. I knew Asterisk was supposed to support it. I figured it out and will show my work here.

The reason is if you have multiple targets (salespeople, doctors, etc), and a call comes into reception. If you only have one parking lot, the receptionist parks the call and gets a parking spot and announces it. If the target cannot be found before the call reverts, the receptionist may park the call again, but may get a different parking spot. The target now has to be told of the new spot.

I think a better approach is to give each target their own parking lot. Then, when a call comes in, the receptionist parks it in the target's parking lot and announces that. The target then will always know where their call is parked. If it reverts, the receptionist can send it back to the same lot.
I implemented 64 parking lots but have cut it down for this posting. In the following example, there are 8 parking lots, each holding up to 9 calls. The parking lots are numbered 22, 23, 24, 25, 26, 27, 28 ,29. To park a call, transfer it to the extension 222, or 223, or 224 etc. To retrieve the call, you dial 422, 423, 424 etc.

There are two files in /etc/asterisk you need to have. You can either login as root and edit them, or, I think there is some sort of FreePBX module that allows editing of such custom files.

The file features_general_custom.conf
================================================================
[parkinglot_22]
context => parkedcalls
parkext => 222
parkpos => 7221-7229
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_23]
context => parkedcalls
parkext => 223
parkpos => 7231-7239
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_24]
context => parkedcalls
parkext => 224
parkpos => 7241-7249
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_25]
context => parkedcalls
parkext => 225
parkpos => 7251-7259
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_26]
context => parkedcalls
parkext => 226
parkpos => 7261-7269
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_27]
context => parkedcalls
parkext => 227
parkpos => 7271-7279
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_28]
context => parkedcalls
parkext => 228
parkpos => 7281-7289
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes

[parkinglot_29]
context => parkedcalls
parkext => 229
parkpos => 7291-7299
parkext_exclusive=yes
parkingtime=20
comebacktoorigin=no
parkinghints=yes
================================================================
and the file extensions_custom.conf
================================================================
[park-return-routing-custom]

exten => _7NNZ,1,Goto(park-return-routing,700,1)

[park-hints-custom]

exten => _4NN,1,Macro(parked-call,,parkinglot_${EXTEN:1})
exten => _4NN,hint,park:7${EXTEN:1}1@parkedcalls&park:7${EXTEN:1}2@parkedcalls&park:7${EXTEN:1}3@parkedcalls&park:7${EXTEN:1}4@parkedcalls&park:7${EXTEN:1}5@parkedcalls&park:7${EXTEN:1}6@parkedcalls&park:7${EXTEN:1}7@parkedcalls&park:7${EXTEN:1}8@parkedcalls&park:7${EXTEN:1}9@parkedcalls
================================================================
Each target can have up to 9 calls parked, and, will be able to see if there is anyone there with a busy lamp field on the 4nn extension.

Outlook Integration

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Does anyone know of a good Outlook integration tool for Freepbx/Asterix? We have tried ADAT but it does not do everything we would like it to.

SIP custom fields

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Hi all,

I'm trying to use an asterisk PBX to connect devices using a customized SIP version.

This devices use additional field both within INVITE method and 200OK.

I've experienced that asterisk act as a back to back entity and so re-generates the SIP messages from zero, removing all the custom fields.

Moreover I've used SIPAddHeader and I was able to copy and paste the custom fields within the INVITE Method... but my connections con't comes up because additional headers are "removed" from 200OK message. So I guess SIPAddHeader is not the right way.

Is there a way to make these message pass through asterisk as they are? Moreover is MESSAGES not managed by asterisk?

Thank You in advance.

Sip Extension As trunk

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Hello,

this my first post , i'm asking if there is method to configure sip extension as trunk .
the scenario i have to dial extension number '804' to get pstn line. i would like to create trunk for this extension

is it possible ?!

Problem with timezone settings wiping out users voicemail settings

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I posted this in general help yesterday and think it was the wrong place. My apology if I've done anything against forum etiquette by re-posting here.

Hi,

I'm attempting to set the voice mail for a couple extensions for different time zones
If I make an extension, then go to voicemail admin and add a time zone to the extension the user name is deleted, and I get '>' in its place also, the voicemail password becomes a '.', the email address a '>', the pager address a '>' and all of the email attachment, play CID, play envelope go blank from a yes choice.

PBX Firmware: 5.211.65-9
PBX Service Pack: 1.0.0.0

Does anyone have any ideas how to fix this?

Thank you

Manual asterisk version update

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Did anyone make or know can I update asterisk verion from 1.8 to 11 use rpm or yum package?

My FreePBX version 211.
Asterisk version 1.8.25.


howto add old caller to agent ?

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hi.
let's ask my question by example :
i have call center for support in our company. there is some customer which calls to get support 2 times in one day to get support. when customer calls for the second time, is it possible to redirect customer's call to the agent which answered for the first time to him???

thanks.

Default route to multiple extentions

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Hi,

i´m new to FreePBX, so far we managed to get most things working as expected.

Comming closer to the point where we have to migrate from our current system to the new one, we are faceing the work to create all inbound routs for every single user.

Is there any way to define a route that catches all incomming calls and transfers them to the extentions refered to with the last digits of the number?

like someone calls 555-1234-30 and is mapped to the user with extention 30 but someone calling 555-1234-40 is mapped to the user with extention 40.

we sure could set up routes for every single user but this apears to be such a common usecase to us, that we belive there musst be a build in way to achive this.

thanks for advice.

Alex

MemberName in Queue

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Hello!

I have updated recently major asterisk version from 1.8 to 11.
In queue_custom_general.conf I set new option log_membername_as_agent = yes.

I use dynamic agents and manage they via dialplan
AddQueuMember((500,${CUT(CHANNEL(name),-,1)},,,${name},${CUT(CHANNEL(name),-,1)})) here variable ${name} is MemeberName.

After that I saw in queue_log 1397044861|1397044860.22540|500|SIP/2516|ADDMEMBER|

Why Memebername value didn't write in queue_log correctly?

I have tested it in another server and saw
1397043506|1397043504.4|500|112|ADDMEMBER|

Unavailable voicemail choice

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How would I go about setting up a dial plan that if no one in a ring group answers an extension freepbx asks the caller to pick who they want to leave a message for. So ... " No one is available to take your call. To leave a message for Tom press 1. To leave a message for Dick press 2. To Leave a message for Harry press 3. Or to leave a general message wait for the tone."

Do I need to use the IVR module?
Any help would be appreciated.

Outbound call center stats?

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I'm looking for a module, of sorts, that will display outbound statistics. I have two call centers - one inbound, the other outbound. QXact would work great for the inbound call center, but not for the outbound call center. Mainly the outbound call center we are just looking for how many calls, by agent, and perhaps average length of call.

Has anyone seen a module like this before?

Separate/Block Numbers of different company sector

Raspbx Freepbx gsm Gateway Detailed Tutorial needed

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i am new to freepbx i intend to buy Goip GSM Gateway then i saw Raspbx Chan Dongle GSM Gateway
so i bought all meterials and almost build halfway and now i am stuck unknowing the FreePBX setup

so far incoming and forwading SMS(http://raspbx/sms/) only works
Raspbx Asterisk (Ver. 11.8.0)

inbound Routes name : inbond

1. inbond Set Destination : Extensions <200> 200-ext
2. Goog motif Set Destination : Extensions <200> 200-ext

outbond Route name : out_GSM

Trunk Sequence for Matched Routes 0 : Dongle (Unlocked E303s-1)
Dial Patterns that will use this Route : NXXNXXXXXX & NXXXXXX
Optional Destination on Congestion : Normal Congestion

Trunks Name : Dongle
Maximum Channels : 1
Custom Dial String : dongle/dongle0/$OUTNUM$

i did spend a lot of time on
http://www.freepbx.org/forum
http://nerdvittles.com/
http://www.raspberry-asterisk.org/ does not give me any clue
http://sourceforge.net/projects/raspbx/

aybody give me better detailed setup for Raspbx GSM Gateway

Thank you


How to activate and diactivate DND from calling from a DID

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Hello,

I am new here,

I need an information how can i do this:

Extentions: 101,102,103,104 etc (all the calls are diverted to their mobiles so users are always offline)

DID NUMBER: 0123456

I want that when the user 101 calls the DID 0123456 the system asks which Extention do you have? the user digit 101 and then the system asks if he want to deactivate or activate the DND.

Please let me know how can i do this.
Thank you.

P.S.
Of course the audio file i will create them.

Fix Outbound calls drop after 30 seconds

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I've run into this twice now on two different routers. The Sonicwall TZ170 and another Zyxel model.

On an intermittent basis, outbound calls that route through these firewalls (and probably others) would simply drop after 30 seconds or so of successful two way audio.

the issue turned out to be a default UDP timeout on the router. Changing the default from 30 seconds to 90 solved the problems.

Interestingly, with one of the Sonicwall Tz170, after a power drop and reboot of the router the default setting was replaced, causing the problem to resurface. Once this was once again changed the calls were no longer dropped.

Dynamic Queue members ring order

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My customer has 8 users that they would like to ring in a linear fashion so every call starts with the same phone and goes through the list. (user 1, user 2, user 3, user 4, user 5, user 6, user 8, user Dirol This works great when the agents are static, however the customer would like to be able to log off agents so calls do not ring a phone that does not have a user attending it. I do not see a ring strategy that will achieve both of these requests. Currently I have assigned an asterisk penalty to make this work with dynamic agents but if an agent is logged in and misses the call it will not move to the next penalty user.

receive fax over fxo gateway

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Hi people,

I have a FreePBX that receive faxes from an analog PBX, this works very well. Recently the FreePBX's pci card died, so I have to replace it.

I don't know if I have to use a PCI card with FXO modules or an ATA gateway with FXO ports. I think that use an ATA gateway is going to be more easy to configure, but I don't know if this configuration has drawbacks.

does anyone have experience with this? Any tip you can give me?

Thanks in advance.

Elastix 2.3.0 , Outbound Calls are getting Blank after 30min

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