Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered before priority 3 callers, for instance ?
I can think of several solutions but none really pleases me :
1. Have 4 different queues, set penalty value and let each caller enter one queue depending on its own priority. I don't like this solution because I foresee editing stats for 4 queues instead of one is harder.
I am in need of a "special" asterisk conference room with the following constraints:
- there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time...
Any hints on how to realize that are highly appreciated..
Thanx in advance, yves
When a call comes in and is parked using the Park() command it appears that
the billing seconds do not include the time while the caller was parked. Is
there a simple way to correct this with an option or setting in asterisk.
This is not acceptable as we are loosing min when callers park calls. I am using 1.8.x release currently. Thanks
I have two asterisk boxes, one with asterisk 220.127.116.11 and the other
with asterisk 18.104.22.168 Sip show settings of both boxes have no difference and also the peers I am generating a call using call file with following details:
Account: 9164421122 < -- this is the accountcode of 1028
CallerID: "Zohair Raza"<1031> < -- I want to see this caller id at
dialing peer (1028) and "Test" <1028> (originiating caller id) at
Hello, I have an issue I remember seeing a while ago and forgot to investigate further. Now it is turning into an issue and will need to be resolved. A customer has Polycom 335 phones (and a couple Soundstation 6000s), and when an extension is calling out, the screen on the 335 shows the company's internal CID number instead of the person they are dialing. This also applies to receiving calls - the internal CID is displayed as opposed to who was calling. I remember seeing something about connectedline issues with Polycom phones, but I can't find the bug I…
In essence Caller ID ETSI and FSK US (Bellcore) is based on the same pattern as;
_____________ ___________________ _________ ________________ _____________
|First Ring burst |_500ms_|Channel seizure 300 bits|__|Mark Signal|__| Caller ID Message|_200 ms_|Second ring burst | So basically any kind of device should be work without any problem, unfortunately during these process if some noises (as miss ground connection or others) happens during the process can make failed to process caller-id information, by the modem. Mc GRATH Ricardo
Thanks Mitul :)
The patch on the link is so old (2006-2007) so I think it's already
implemented in the newest version. Honestly to say, I already try any
combitions but still the caller id doesn't work :( cidsignalling=bell,dtmf,v23
cidstart=ring,polarity,dtmf with some parameter if we set it to dtmf Hopeless :(( On Sun, Jun 3, 2012 at 4:51 PM, Mitul Limbani
> Look at this issue : https://issues.asterisk.org/view.php?id=6683
> And try different combinations suggested over there, you might get lucky…
In order solve my incoming caller ID problem, I upgrade the dahdi to
version 2.6.1 from version 2.4.x. After upgrade, I found the echo
cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I
can hear my self talking on the phone. How to solve this? I think I need to
recompile dahdi 2.6.1 with OSLEC support? how? [root@callcenter ~]# dahdi_cfg -vvv
DAHDI Tools Version - 2.6.1 DAHDI Version: 2.6.1
Echo Canceller(s): HWEC
Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01)