How to configure 2 or more user to call a single conference extension

Asterisk Asterisk Support
1

Hello Everyone,

Problem : How to configure 2 or more user to call a single conference extension

We have Asterisk 16.9.0 installed in a CentOS 7 server at our office’s local network. The only configurations we have made on this server are those we are making mention here.

–>What we want to do:

We want to configure such that, our Mobile network callers, should call our DID number and it should lead them to a conference through a single extension, which carries our DID number as the number to be called for the conference. We have configured 2 kinds of users in confbridge.conf: Admin and guest user with PINs. We want that the mobile network callers should all call the DID but they should enter the PIN of the guest user which we will give them and the admin will enter the PIN of the admin user account in confbridge.conf we will give them.

–>What we have been able to do:

We have been able to configure our system such that, our mobile network callers, call the DID and they are asked to put the PIN of the conference. But according to what we have been able to do, only the admin PIN functions, because it is it that is in our Dial plan.

Our configurations are below:

In confbridge.conf


-----------User accounts------------

[wv_admin_user_DID]
type=user
pin=6666
marked=yes
admin=yes
music_on_hold_when_empty=yes
announce_user_count=yes
announce_only_user=yes
announce_join_leave=yes

[wv_guest_user_DID]
type=user
pin=4444
wait_marked=yes
end_marked=yes
music_on_hold_when_empty=yes
announce_user_count=yes
announce_user_count_all=yes
announce_only_user=yes
announce_join_leave=yes

---------- ConfBridge Bridge------------

[wv_bridge_DID]
type=bridge
max_members=50
video_mode = follow_talker

We didn’t do anything the at level of MENU

In sip.conf


[general] 
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
canreinvite = no
dtmfmode=auto

register => sipusername:passwd@xxx.xx.xxx.xx:5060/+237222518004
[sipusername]
type=peer
context=trunkinbound
secret= passwd
host= xxx.xx.xxx.xx
nat=force_rport,comedia
fromdomain= xxx.xx.xxx.xx
fromuser=sipusername
username=sipusername
insecure=invite
disallow=all
allow=ilbc
allow=g726
allow=gsm
allow=g723
allow=ulaw
dtmfmode=inband
quality=yes


[201]
user=201
type=friend
host=dynamic
disallow=all
allow=ulaw
secret=201
context=trunkinbound
nat=comedia

In extensions.conf

exten => ourusername,1,Dial(SIP/ourusername)

exten => 201,1,Dial(SIP/201,20)
exten => 201,2,Answer()
exten => 201,3,Playback()
exten => 201,4,Hangup()


exten => +237222518004,1,Progress()
exten => +237222518004,2,Wait(1)
exten => +237222518004,3,ConfBridge(1,wv_bridge_DID,wv_admin_user_DID)

This above extensions.conf what we put which only considers the admin PIN. We agree it is normal because only it, is mentioned in the dial plan.

In attempt to advance to our need, we changed our extensions.conf as follow:

[trunkinbound]

	exten => +237222518004,1,NoOP()
	same => n,Answer()
	same => n,Set(CONFBRIDGE(wv_guest_user_DID,wv_admin_user_DID)=yes)
	same => n,ConfBridge(1,wv_bridge_DID)
	same => n,Hangup()

	exten => ourusername,1,Dial(SIP/ourusername)


	exten => 201,1,Dial(SIP/201,20)
	exten => 201,2,Answer()
	exten => 201,3,Playback()
	exten => 201,4,Hangup()

Thank you for your quick response. We appreciate.

2

Not answering your question, but this is not a valid combination!

3

Thank you david551.
It works for us but i will try removing it and see what it gives and give you the feedback.

4

I have removed :point_down:

allow=gsm

And it still working fine.

Thank you.

5

Have you tried having ONE extension to dial for external users, and one for internal, then configure them with different user profiles? Like this:

exten => +237222518004,1,Progress()
exten => +237222518004,2,Wait(1)
exten => +237222518004,3,ConfBridge(1,wv_bridge_DID,wv_guest_user_DID)

exten => 1234,1,Progress()
exten => 1234,2,Wait(1)
exten => 1234,3,ConfBridge(1,wv_bridge_DID,wv_admin_user_DID)

The 1234 extension needs to be places in the context where your internal extensions lives, and the external number in whatever inbound context might be appropriate.

6

Thank you very much for your intervention sir.

All our users will be external.

7

So all will use the +237 number?

In that case, you can either implement the PIN checking yourself, by disabling PIN on the conference user, then request the PIN and select the correct user profile, based on the PIN entered, or you can have two DID numbers, one for admin users, one for guests.

Alternatively, you can build a menu where you select either guest or admin before connecting to the conference, and entering the PIN, then the menu can set the profile.

But from what I can see, there’s no way to select the profile, based on the PIN entered.

8

So all will use the +237 number?

Yes Sir.

Please if you have resources help me with so i can go and study. Or help me with a sample call forwaring example for me to follow.

I have been reading from different websites but …

Thank you very much for your quick response.

9

I have 2 DIDs linked to my SIP Account. And i configured them to 2 users (Each DID for each user account) of same conference, but when i test them by calling, they all fall on the first number.

10

Sounds wrong, what does the dialplan and sip.conf look like? (Remember to mask sensitive information like passwords etc.)

Also you can try to put
same => n,DumpChan()
Somewhere in the beginning of your extension configuration, to see what DID Asterisk sees.

11

Thank you for the follow up.

My boss has demanded that we implement IVR to redirect the calls to their various segments. So this issue is no more a priority for us.
Thank you all for helping out.

12

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