Call limit for Hunt Group

Asterisk Asterisk Dialplan
1

Hi guys,
I need a help with this problem:
I have a hunt group in extensions.conf:

exten => 2000,1,Dial(SIP/101&SIP/107&SIP/109)

So whenI dial the 2000 call will ring on all of this three peers.

in sip.conf i have defined sip peers 101,107,109

And I would like to do this:
When all of this three peers has a connected call, the fourth call will get waiting music and must wait when one of this trhee peers finish active call.

Is that possible to do?
Thanks for response
(And sorry for my english :slight_smile: )

2

Use the Queue application, instead.

1 Like
3

Thanks David,
I tried read something about. So I found how to set up queue.conf but it still not works for me.
Probably I misunderstood how to set extensions.conf to make it works like I described in article above.
I set this:

queue.conf
-----------
[general]
persistentmembers=yes

[fronta]
ringinuse=no
musicclass=default      ; play [default] music
strategy=ringall       ; use the Round Robin Memory strategy
timeout=30
context= promo
member=> 101/promo,1
member=> 107/promo,1
member=> 109/promo,1

extensions.conf
---------------------------
exten => 2000,1,Queue(fronta)

sip.conf
-----------
[101]
type=friend
context=promo

host=dynamic
username=101
secret=101
callerid=2000
callcounter=yes
qualify=yes


[107]
type=friend
context=promo

host=dynamic
username=107
secret=107
callerid=2000
qualify=yes
callcounter=yes


[109]
type=friend
context=promo

host=dynamic
username=109
secret=109
callerid=2000
qualify=yes
callcounter=yes


...and etc.

Any help please?

4

You have the wrong strategy for a simple hunt group.

Your queue member names don’t have a technology, and don’t match the syntax for any likely technology. The sample configuration shows you what they should look like.

5

Thanks, now it works.
Or more precisely: Now Customers can call number 2000 and they reach my three agents and they start rings.
Now the question: When customer wait for pickup by agent, he didn´t hear anything. And I would like to hear normaly beeping or music on hold, or anything except silence.
How can I do that?

6

If you want ringing rather than MOH, you need to enable it in the options. Calling Ringing() first may also happen.

For music on hold, you need to call Progress, if your provider supports early media, or Answer the call, together with properly configuring at least the default music on hold.

7

Thanks, when I´m put into extensions.conf

exten => 2000,1,Ringing()

it rings to the customer! :smiley:

But, new problem:
I would like to set the announcements.

So I set this:

queue.conf

[general]
autofill=yes             
shared_lastcall=yes      

[StandardQueue](!)      
strategy=ringall       
joinempty=yes            
leavewhenempty=no       
wrapuptime=10
timeout=30
announce-frequency=30          
min-announce-frequency=30    
periodic-announce-frequency=45  
random-periodic-announce=no  
relative-periodic-announce=yes  
announce-holdtime=once      
announce-position=limit      
announce-position-limit=10     
announce-round-seconds=30      
min-announce-frequency=30      
periodic-announce-frequency=45  
random-periodic-announce=no
relative-periodic-announce=yes  
announce-holdtime=once         
announce-position=limit       
announce-position-limit=10     
announce-round-seconds=30 


[fronta](StandardQueue)
musicclass=default
ringinuse=no
member => SIP/101

extensions.conf

exten => 2000,1,Verbose(2,Joining the support queue for a maximum of 2 minutes)
exten => 2000,n,Queue(fronta,120)
exten => 2000,n,VoiceMail(2000,u)
exten => 2000,n,Hangup()

and asterisk sends me this:

asterisk*CLI>
  == Using SIP RTP CoS mark 5
       > 0x7f59180067d0 -- Strict RTP learning after remote address set to: 10.48.67.120:41000
    -- Executing [2000@promo:1] Verbose("SIP/103-00000006", "2,Joining the support queue for a maximum of 2 minutes") in new stack
  == Joining the support queue for a maximum of 2 minutes
    -- Executing [2000@promo:2] Queue("SIP/103-00000006", "fronta,120") in new stack
    -- Started music on hold, class 'default', on channel 'SIP/103-00000006'
  == Using SIP RTP CoS mark 5
    -- Called SIP/101
    -- SIP/101-00000007 connected line has changed. Saving it until answer for SIP/103-00000006
    -- SIP/101-00000007 is ringing
       > 0x7f5954006f80 -- Strict RTP learning after remote address set to: 10.48.67.64:10022
    -- SIP/101-00000007 connected line has changed. Saving it until answer for SIP/103-00000006
    -- SIP/101-00000007 answered SIP/103-00000006
    -- Stopped music on hold on SIP/103-00000006
    -- Channel SIP/101-00000007 joined 'simple_bridge' basic-bridge <07dea475-23e3-47e4-9b62-68bfb011137b>
    -- Channel SIP/103-00000006 joined 'simple_bridge' basic-bridge <07dea475-23e3-47e4-9b62-68bfb011137b>
       > 0x7f5954006f80 -- Strict RTP switching to RTP target address 10.48.67.64:10022 as source
       > 0x7f59180067d0 -- Strict RTP switching to RTP target address 10.48.67.120:41000 as source
       > 0x7f59180067d0 -- Strict RTP learning complete - Locking on source address 10.48.67.120:41000
       > 0x7f5954006f80 -- Strict RTP learning complete - Locking on source address 10.48.67.64:10022
asterisk*CLI>
asterisk*CLI>
  == Using SIP RTP CoS mark 5
       > 0x7f591801ab30 -- Strict RTP learning after remote address set to: 10.48.67.116:41000
    -- Executing [2000@promo:1] Verbose("SIP/104-00000008", "2,Joining the support queue for a maximum of 2 minutes") in new stack
  == Joining the support queue for a maximum of 2 minutes
    -- Executing [2000@promo:2] Queue("SIP/104-00000008", "fronta,120") in new stack
    -- Started music on hold, class 'default', on channel 'SIP/104-00000008'
    -- Stopped music on hold on SIP/104-00000008
    -- <SIP/104-00000008> Playing 'queue-youarenext.gsm' (language 'cs')
    -- Told SIP/104-00000008 in fronta their queue position (which was 1)
    -- <SIP/104-00000008> Playing 'queue-thankyou.slin' (language 'cs')
    -- Started music on hold, class 'default', on channel 'SIP/104-00000008'

But I still hear nothing. How can I solve this if Asterist didn´t sent me back any err or warn message?

I try copy original (en) files into my cs folder but still didn´t work

8

Your service provider probably doesn’t support early media from end users (it would allow people to get free information service calls). You will need to Answer the call before calling Queue.

(Even if they do support early media, you may need to call Progress(); I’ a little confused as to whether that is really necessary.

1 Like