Xlite Btw, how to find out which codec a call is using when asterisk is dialing out ?
On Thu, Apr 16, 2009 at 11:05 PM, David @ULC <[email protected]> wrote: > > Which is the latest version of Asterisk ? > > > On Thu, Apr 16, 2009 at 11:04 PM, David @ULC <[email protected]> wrote: > >> busy-level ? >> >> How to use it and whats the purpose ? >> >> >> On Thu, Apr 16, 2009 at 10:43 PM, David @ULC <[email protected]> wrote: >> >>> >>> http://threebit.net/mail-archive/asterisk-users/msg07138.html >>> Remember that if you want to support attended transfers, you need at >>> least two >>> simultaneous calls. >>> >>> So, its safe bet to keep call-limit=2. >>> >>> Advice ? >>> >>> >>> On Thu, Apr 16, 2009 at 10:37 PM, David @ULC <[email protected]>wrote: >>> >>>> My SIP config is below : >>>> >>>> [sip64] >>>> type=peer >>>> username=fiduci >>>> fromuser=fiduci >>>> authuser=fiduci >>>> secret=pass >>>> host=64.33.22.11 >>>> nat=no >>>> canreinvite=yes >>>> insecure=very >>>> disallow=all >>>> allow=g729 >>>> allow=ulaw >>>> context=default >>>> dtmfmode=rfc2833 >>>> >>>> Now, I need to add another element as call-limit=1 and this should >>>> solve my problem ? >>>> >>>> If yes. Great. Kindly advice. >>>> >>>> But will that allow 3 party conference ? >>>> >>>> >>>> On Thu, Apr 16, 2009 at 10:22 PM, David @ULC <[email protected]>wrote: >>>> >>>>> "call-limit in sip.conf" >>>>> >>>>> Can you elaborate please and how to set that. >>>>> >>>>> Lets presume I have 10 agents and dial ratio is 4. >>>>> >>>>> >>>>> On Thu, Apr 16, 2009 at 10:06 PM, David @ULC <[email protected]>wrote: >>>>> >>>>>> >>>>>> Even I thought so thats why I tried with 4 VOIP provider and things >>>>>> didn't change. :-( >>>>>> >>>>>> >>>>>> On Thu, Apr 16, 2009 at 8:36 PM, David @ULC <[email protected]>wrote: >>>>>> >>>>>>> >>>>>>> >>>>>>> Many time we face an issue where even if an agent is on Call, another >>>>>>> call comes in. >>>>>>> >>>>>>> Sometimes, even if agent hang up the call, call stays back and >>>>>>> another come sin and then both customers can hear each other { which i >>>>>>> think >>>>>>> is VERY dangerous [image: Wink] } >>>>>>> >>>>>>> Also, this thing happens even when we have just 5 agents on a single >>>>>>> server. [image: Sad] >>>>>>> >>>>>>> Our version is Asterisk 1.2.27 >>>>>>> >>>>>>> Any Solutions ? >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>> >> >
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
