Re: [Asterisk-Users] Call Center Phone with Auto Answer
Hi Steve ,I was actually looking forward for the same thing , do y ou have something like this , as an example?regardsJunaid UppalOn 5/9/06, Steve Totaro [EMAIL PROTECTED] wrote: Use an activex screenpop.Thanks,Steve Totaro -Original Message- From: Kevin Savoy [mailto:[EMAIL PROTECTED]] Sent: Monday, May 08, 2006 3:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer This may be the way to go but not the best. Our agents frankly aren't the brightest people and I can see them forgetting it as soon as it issaid to them, or they are not paying attention and missing the announcementbut it is something to look into. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of TimeBandit Sent: Monday, May 08, 2006 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Ok I can get this to work now the next problem is since the agentstays off-hook when a call is presented to them there is no indicationof what call this is. Being an inbound call center we have 100's of clients. 1,000's of toll frees and DNIS. We use the Asterisk callerID function toassign a name to each call so that when the call is presented to the agent it displays which company the call is for. With AgentLogin all the agent gets is the number they dialed to log in. No idea which client this callis for. Any ideas there? When you send the caller to the queue, you can pass the name of the audio file to be played as the announcement to the agent when he gets the call. Maybe you could use that and pre-record the name of the customer, passing that audio file something like exten = 8000,n,Queue(8000|t||FilenameOfTheCustomerName|300) maybe you could also use festival hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need Help on Areski Calling Card Solution plz
AreskiCC works great for me , i've been using it for ~ 500 + cards scene and it works awesome for me! really , the guy did a REALLY good job , trust me. cheers ~uppal On 9/25/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi:My experience with Areski is I wasn't able to get itto work and wasn't able to get help including from the owner of idiot guide who inturns wasn't able to getareski to work either according to him.I easily downloaded astcc and works fineRegards;Chawki Hammoud--- ADEGOKE ARUNA [EMAIL PROTECTED] wrote: Can someone share its working files experience on areskicc with me. I got it installed but my sip user and iax could not get registered talkless of making call and all the include directives instructed in the idiot guide were followed. Can someone share its experience with me on this? Aruna -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of CM Rahman Jr. Sent: Tuesday, July 19, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Comments on Areski Calling Card Solution plz I am using it. I liked it. The guy did a good job. He doesn't have the agent module yet. But I think that is on its way. Thanks Quoting Arnd Vehling [EMAIL PROTECTED]: Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection around http://mail.yahoo.com___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AreskiCC + Mutliple SIP Gateways for one route
Hello There, I'd like to define multiple providers for one dial prefix , like , i want if my one trunk gateway is filled the call is transfered to other ip, how can i achieve it with areskicc.Kindly Help. cheers Thanks Junaid Uppal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AreskiCC Calling Problem
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it tells me i have ten $ , right after that it says sorry you
dont have enough funds for this call and hangs up. i see this in cli
help me out please guys , thanks a lot!!
regards
~junjun
--
CLI LOG START
--
areskicc2.php: 'agi_callerid' = '1001'
areskicc2.php: 'agi_calleridname' = 'Junaid Uppal'
areskicc2.php: 'agi_callingpres' = '0'
areskicc2.php: 'agi_callingani2' = '0'
areskicc2.php: 'agi_callington' = '0'
areskicc2.php: 'agi_callingtns' = '0'
areskicc2.php: 'agi_dnid' = '011905'
areskicc2.php: 'agi_rdnis' = 'unknown'
areskicc2.php: 'agi_context' = 'default'
areskicc2.php: 'agi_extension' = '011905'
areskicc2.php: 'agi_priority' = '3'
areskicc2.php: 'agi_enhanced' = '0.0'
areskicc2.php: 'agi_accountcode' = ''
areskicc2.php:
areskicc2.php: ANSWER
areskicc2.php: string(48) 1001 ; SIP/1001-d6fb ; 1118521907.13 ; ; 011905n
areskicc2.php: string(26) Requesting DTMF :: Len-10n
areskicc2.php: GET DATA prepaid-enter-pin-number 1 10
-- Playing 'prepaid-enter-pin-number' (language 'en')
areskicc2.php: string(21) RES DTMF : 5882431851n
areskicc2.php: string(25) CARDNUMBER :: 5882431851n
areskicc2.php: string(94) SELECT credit, tariff, activated, inuse,
simultaccess FROM cc_card WHERE username='5882431851'n
areskicc2.php: array(1) {n [0]=n array(5) {n[0]=n
string(2) 10n[1]=nstring(1) 1n[2]=nstring(1)
tn[3]=nstring(1) 0n[4]=nstring(1) 0n }n}n
areskicc2.php: STREAM FILE prepaid-you-have #
areskicc2.php: SAY NUMBER 10 X
-- Playing 'digits/10' (language 'en')
areskicc2.php: STREAM FILE prepaid-dollars #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse+1 WHERE
username='5882431851'n
areskicc2.php: CHANNEL STATUS SIP/1001-d6fb
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: STREAM FILE prepaid-no-enough-credit-stop #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse-1 WHERE
username='5882431851'n
areskicc2.php: STREAM FILE prepaid-final #
-- AGI Script areskicc2.php completed, returning 0
-- Executing Wait(SIP/1001-d6fb, 2) in new stack
-- Executing Hangup(SIP/1001-d6fb, ) in new stack
== Spawn extension (default, 011905, 5) exited non-zero on 'SIP/1001-d6fb'
-
CLI LOG ENDS
here's the /tmp/areskicc-errors.log
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[TRY :
callingcard_ivr_authenticate]
[11/06/2005
16:08:51]:[CallerID:1001]:[CN:5882431851]:[callingcard_acct_start_inuse]
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse+1 WHERE username='5882431851']
[11/06/2005 16:08:51]:[CallerID:1001]:[CN:5882431851]:[CHANNEL STATUS : 6]
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[Start: UPDATE
cc_card SET inuse=inuse-1 WHERE username='5882431851']
[11/06/2005 16:08:53]:[CallerID:1001]:[CN:5882431851]:[exit]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] Radius on *
Hey I am intrested in testing this thing out , i was already trying to work out something of my own when i found this ,s o can we work on this? ~uppal On Wed, 19 Jan 2005 11:23:26 +0200, Mike Tkachuk [EMAIL PROTECTED] wrote: Hello All, Want to say that it not latest version of this tool. Currently I created project on berlios.de called b2bua. There I will post latest version. What is this tool? Currently it's AGI script with many asterisk patches. It support some authorization, accounting (start, stop), LCR, 'Smart' failover support. Currently it include few asterisk patches and applications. It developed to use with SIP protocol, but I think with small changes it will work with any other. But I think I need some help from peoples to test it. I do not promise nothing, will see what will be next. All who interesting in this development or testing, please write to this thread. __ Mike Tkachuk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadbri NT_mode S-Bus Problem
Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from
Siemens Switzerland , What I've done is to get one cable from ISDN NT
-- ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then
used bristuff ( google for it ) , and used that , it just works! . I
can send you my configs if you need som ehlp
regards
~uppal
On Sun, 18 Jul 2004 23:47:47 +0200, Ben Bosshardt
[EMAIL PROTECTED] wrote:
What type is your ISDN house telephone system?
Without more specific information all we can do is guess...
Our system is a just the basic subscription to SWISSCOM, which is the main
phone company in Switzerland. We have BRI with 2 Channels which can be used
simulaniously and a Siemens NT that has only the function of feeding our
S-bus with 4 phones connected.
For a sollution to 1 ... drop the r option of dial...
exten = _X.,1,Dial(Zap/g1/${EXTEN})
I will give it a try.
You might need pridialplan/prilocaldialplan set to local for local
calls... or both to unknown... just experiment with those values.
I am still looking for any documentation regarding the use of
pridialplan/prilocaldialplan. I don't know how to find out what SWISSCOM
requires.
Thanks for your help.
Ben
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hotline
Hello There, I tried checking out for this feature , what i want to do is that as soon as the user picks up the handset , * waits for 10 secs and then dials a predefined number , its like the HOTLINE feature we have in normal POTs . Is it possible with Asterisk? If yes then how? Regards ~uppal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] random disconnect with hfc ISDN card and sipura
Tomaz, I am using HFC ISDN card too , without any problem , I have a cisco phone and my call never gets disconnected . Regards -junaid p.s: how did you compile bristuff with latest cvs? On Thu, 15 Jul 2004 15:01:30 +0200, Tomaz [EMAIL PROTECTED] wrote: hello asterisk people ;) i have a problem disconecting me if i talk to someone thru isdn hfc based card (from sip phone sipura 2000 to telco), i get diconnected in 2-4 minutes randomly .. ? anyone has a same problem ? where to look? - latest asterisk CVS + bristuff 0.1.0 same with 0.0.2 .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
