[asterisk-users] Best way to limit outgoing calls per trunk

Vardan Harutyunyan hvardan71 at gmail.com
Mon May 31 20:38:52 CDT 2010


A ok, I think I have understand what you want.
The first, are you want that a2b calculate the buying price?
If it for you not so important, the you can use failover trunk in a2b.
Try this.
If no, then you can you dialplan to explain what he must do on hangup cause.

I use AEL. For example,

	Dial(SIP/${AGENTSPHONE});
	Noop(${DIALSTATUS});
        switch(${DIALSTATUS}) {
                     case BUSY:
                         Noop(================ Busy);
			Playback(${AGENT_ALLBUSY_MESSAGE});
                         break;
                     case CHANUNAVAIL:
                         Noop(================ Channel Unavailable);
			Playback(${AGENT_UNAVAILABLE_MESSAGE});
                         break;
                     case NOANSWER:
                         Noop(================ No answer);
			Playback(${AGENT_ALLBUSY_MESSAGE});
                         break;
                     case CANCEL:
                         Noop(================ Cancel);
			Playback(${AGENT_ALLBUSY_MESSAGE});
                         break;
                     case CONGESTION:
                         Noop(================ Congestion);
			Playback(${AGENT_UNAVAILABLE_MESSAGE});
                         break;
                     case ANSWER:
                         Noop(================ Answer);
                         break;
                     default:
                         Noop(================ Default);
			Playback(${AGENT_UNAVAILABLE_MESSAGE});
                         break;
                 };
-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: info at eif.am
www.eif-it.com

bruce bruce wrote:
> Hi Vardan,
>
> I am using use_dnid=yes and then setting the Account Code in Asterisk
> dialplan before sending the call to A2Billing _x. context which
> automatically dials. So, before the call goes to A2Billing, I can check
> to see if there is a channel up or not. I am not sure how the local
> channel you mentioned works. Would appreciate it if you share.
>
> Can you determine the number of channels in the queue?
>
> One of my trunks allows for 3 calls certain time of the day and sometime
> it allows for only 1 channel. Hence the need for this.
>
> Thanks,
>
>
> On Mon, May 31, 2010 at 11:39 AM, Vardan Harutyunyan
> <hvardan71 at gmail.com <mailto:hvardan71 at gmail.com>> wrote:
>
>     No, if You use call-limit the call will be dropped.
>     How you put your customer on hold?
>     If you use queue and the customer hear the music onhold, he will be
>     billed for this connection
>     I have try use queue and a2b, and I have do all connection using local
>     channel, so I have become all is works, and the customer after speaking
>     with agents and transferred to international number, is billed only for
>     international call.
>
>     Sorry for my english, if any question, please write. I will try to
>     explain.
>

>     Thanks
>
>     --
>     Vardan Harutyunyan,
>     Senior System Administrator
>
>     Enterprise Incubator Foundation
>     123 Hovsep Emin Street,
>     Yerevan 0051, Republic of Armenia
>     Tel: + 374 10 219735
>     Fax: + 374 10 219777
>     E-mail: info at eif.am <mailto:info at eif.am>
>     www.eif-it.com <http://www.eif-it.com>
>
>     bruce bruce wrote:
>      > Thanks for the advice, but I have to keep the customer on hold
>     till the
>      > line becomes available. Is that possible by the method you
>     mentioned? I
>      > am using A2B 1.7 and Asterisk 1.4.
>      >
>      > Thanks,
>      >
>      >
>      > On Mon, May 31, 2010 at 2:27 AM, Vardan Harutyunyan
>     <hvardan71 at gmail.com <mailto:hvardan71 at gmail.com>
>      > <mailto:hvardan71 at gmail.com <mailto:hvardan71 at gmail.com>>> wrote:
>      >
>      >     Hello,
>      >
>      >     What version of Asterisk You are use?
>      >     And what version of A2Billing You are use?
>      >     If You use version 1.4.X of Asterisk You can put call-limit
>     string in
>      >     sip.conf for this trunk
>      >
>      >     If You use A2B ver 1.7 and Asterk 1.4 you can announce this
>     trunk using
>      >     sip config in A2B, and the are call-limit via web.
>      >
>      >     And how I know, in 1.6 is no more call-limit in sip.conf
>      >
>      >
>      >     --
>      >     Vardan Harutyunyan,
>      >     Senior System Administrator
>      >
>      >     Enterprise Incubator Foundation
>      >     123 Hovsep Emin Street,
>      >     Yerevan 0051, Republic of Armenia
>      >     Tel: + 374 10 219735
>      >     Fax: + 374 10 219777
>      >     E-mail: info at eif.am <mailto:info at eif.am> <mailto:info at eif.am
>     <mailto:info at eif.am>>
>      > www.eif-it.com <http://www.eif-it.com> <http://www.eif-it.com>
>      >
>      >     bruce bruce wrote:
>      > > Thanks for that. It very well detailed.
>      > >
>      > > I am not sure if I can use GROUP and GROUP_COUNT now that I see
>      >     how it's
>      > > used. You see, the call is placed by A2Billing so I don't have a
>      >     control
>      > > over setting GROUP increase and so if there is a call GROUP_COUNT
>      >     won't
>      > > work.
>      > >
>      > > I might resort back to using "sed" and "awk" to take output of
>     "core
>      > > show channels" and check for it's state. I will appreciate some
>      >     guru of
>      > > "sed" to to give me a true false for a channel up or not using
>      > "sed" and
>      > > "core show channels"
>      > >
>      > > Thanks,
>      > > Bruce
>      > >
>      > > On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman
>      > > <jonathan at thurmantech.com <mailto:jonathan at thurmantech.com>
>     <mailto:jonathan at thurmantech.com <mailto:jonathan at thurmantech.com>>
>      > <mailto:jonathan at thurmantech.com
>     <mailto:jonathan at thurmantech.com> <mailto:jonathan at thurmantech.com
>     <mailto:jonathan at thurmantech.com>>>>
>      >     wrote:
>      > >
>      > >     On Sun, May 30, 2010 at 9:37 AM, bruce bruce
>      > <bruceb444 at gmail.com <mailto:bruceb444 at gmail.com>
>     <mailto:bruceb444 at gmail.com <mailto:bruceb444 at gmail.com>>
>      > > <mailto:bruceb444 at gmail.com <mailto:bruceb444 at gmail.com>
>     <mailto:bruceb444 at gmail.com <mailto:bruceb444 at gmail.com>>>> wrote:
>      > > > Thanks for the tip. I have been checking those two options. Would
>      > >     you be
>      > > > able to provide an example of how GROUP or GROUP_COUNT may check
>      > >     for a trunk
>      > > > usuage?
>      > >
>      > >     Here is how I do it.  It is based on Asterisk 1.6.1.x, and I
>      >     created a
>      > >     generic sub-routine to call for limiting trunks to a specific
>      >     number
>      > >     of calls.  The code is documented, so it should give you a
>      >     good idea
>      > >     of how to use it.
>      > >
>      > > http://thurmantech.com/node/7
>      > >
>      > >     -Jonathan
>      > >
>      > >
>      > > >From what I see is that you have to assing certain routes a group
>      > > > and then count the group, but how I do include a trunk in the
>      >     group?
>      > > > Thanks
>      > > >
>      > > > On Sat, May 29, 2010 at 7:07 PM, Steve Edwards <asterisk.org
>     <http://asterisk.org>
>      > <http://asterisk.org>
>      > > <http://asterisk.org>@sedwards.com <http://sedwards.com>
>     <http://sedwards.com>
>      > <http://sedwards.com>>
>      > > > wrote:
>      > > >>
>      > > >> On Sat, 29 May 2010, bruce bruce wrote:
>      > > >>
>      > > >> > I am looking to use System() function along with some bash
>      > >     scripting to
>      > > >> > determine if a Trunk is being used during certain time of the
>      > >     day or
>      > > >> > not. Here is what I have in mind. Please guide me if you know
>      > >     a better
>      > > >> > way:
>      > > >>
>      > > >> Using the GROUP/GROUP_COUNT functions in the dialplan is a
>      > >     better way.
>      > > >>
>      > > >> Using system() will mean creating a bunch of processes (each
>      > > >> sed/awk/cut/etc command).
>      > > >>
>      > > >> --
>      > > >> Thanks in advance,
>      > > >>
>      > >
>      >
>     -------------------------------------------------------------------------
>      > > >> Steve Edwards sedwards at sedwards.com
>     <mailto:sedwards at sedwards.com> <mailto:sedwards at sedwards.com
>     <mailto:sedwards at sedwards.com>>
>      > > <mailto:sedwards at sedwards.com <mailto:sedwards at sedwards.com>
>     <mailto:sedwards at sedwards.com <mailto:sedwards at sedwards.com>>>
>      >       Voice: +1-760-468-3867 PST
>      > > >> Newline                                              Fax:
>      > >     +1-760-731-3000
>      > > >>
>      > > >> --
>      > > >>
>      > >
>      >
>     _____________________________________________________________________
>      > > >> -- Bandwidth and Colocation Provided by
>      > > http://www.api-digital.com --
>      > > >> New to Asterisk? Join us for a live introductory webinar every
>      > >     Thurs:
>      > > >> http://www.asterisk.org/hello
>      > > >>
>      > > >> asterisk-users mailing list
>      > > >> To UNSUBSCRIBE or update options visit:
>      > > >> http://lists.digium.com/mailman/listinfo/asterisk-users
>      > > >
>      > > >
>      > > > --
>      > > >
>      >
>     _____________________________________________________________________
>      > > > -- Bandwidth and Colocation Provided by
>      > http://www.api-digital.com --
>      > > > New to Asterisk? Join us for a live introductory webinar every
>      >     Thurs:
>      > > > http://www.asterisk.org/hello
>      > > >
>      > > > asterisk-users mailing list
>      > > > To UNSUBSCRIBE or update options visit:
>      > > > http://lists.digium.com/mailman/listinfo/asterisk-users
>      > > >
>      > >
>      > >     --
>      > >
>      >
>     _____________________________________________________________________
>      > >     -- Bandwidth and Colocation Provided by
>      > http://www.api-digital.com --
>      > >     New to Asterisk? Join us for a live introductory webinar
>      >     every Thurs:
>      > > http://www.asterisk.org/hello
>      > >
>      > >     asterisk-users mailing list
>      > >     To UNSUBSCRIBE or update options visit:
>      > > http://lists.digium.com/mailman/listinfo/asterisk-users
>      > >
>      > >
>      >
>      >
>      >     --
>      >
>     _____________________________________________________________________
>      >     -- Bandwidth and Colocation Provided by
>     http://www.api-digital.com --
>      >     New to Asterisk? Join us for a live introductory webinar
>     every Thurs:
>      > http://www.asterisk.org/hello
>      >
>      >     asterisk-users mailing list
>      >     To UNSUBSCRIBE or update options visit:
>      > http://lists.digium.com/mailman/listinfo/asterisk-users
>      >
>      >
>
>
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>     New to Asterisk? Join us for a live introductory webinar every Thurs:
>     http://www.asterisk.org/hello
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>
>




More information about the asterisk-users mailing list