When a customer calls you back by dialing your GSM gateway (simple chip), is the incoming call being routed to the right person?
Let's imagine the following situation: someone from your company calls a customer to discuss a certain matter, the customer, being unavailable at the moment, does not answer, but then returns the call to the chip number that called them, dialing your GSM gateway , when your telephone exchange receives this call coming through the gateway, who does it forward to? To the secretary, to an URA, to a queue? If so, how will the person answering the phone know who dialed this customer or what the call was about? How much could your company be losing due to a simple detail like this? With this in mind, SipPulse created the ATI EF:
Aiming to optimize the callback process for GSM gateways, SipPulse developed a new service linked to ATI, called ATI-EF (Extension Fidelity), making this facility available to all ATI users, at no cost for activation.
How it works?
Saving the information: When your exchange carries out a portability query on the ATI, it will also inform which extension is dialing this number, then the ATI will save this information for a period of two weeks.
Who called: To find out which was the last extension that dialed this number that is calling your GSM gateway, simply make a new query to ATI, asking which was the last extension that dialed this number, then ATI will inform the extension number and your exchange can forward the call directly to it.
How is the consultation carried out?
This service was designed to be easy to use and is present in both portability queries via SIP Redirect and via WEB Service.
To carry out the portability query and store the extension via WEB Service, the URL to be used is http://port.sippulse.com:9091/?from=RAMAL&num=NUMERO, where EXTENSION is the extension you want to store and NUMBER is the number dialed, which will be consulted for portability, which must be in the DDD+Number pattern (ex 4899998888)
When receiving a call from the previously dialed number and finding out which extension made the first call, the URL is http://port.sippulse.com:9091/?from=NUMBER, where NUMBER is the number of the caller (A number ) and must be in the same pattern as the portability query.
To consult via SIP Redirect, it is necessary to add the X-Extension: EXTENSION header to the SIP protocol, whereas to discover the extension the header is X-Get-Extension.
How to configure in Asterisk?
Via WEB Service
Outgoing call:
[from-internal]
exten => _X.,1,Set(CURLOPT(userpwd)=USER:PASSWORD)
same=>n,Set(operator=${CURL(https://port.sippulse.com:9091/?from=${CALLERID(num)}&num=${EXTEN})})
same => n,NoOP(${operator})
Incoming call:
[from-pstn]
exten => s,1,NoOP(${CALLERID(num)})
same => n,goto(ati-ef,${CALLERID(num)},1)
[ati-ef]
exten => _X.,1,Set(CURLOPT(userpwd)=USER:PASSWORD)
same => n,NoOP(${CALLERID(num)})
same => n,Set(ramal=${CURL(https://port.sippulse.com:9091/?from=${CALLERID(num)})})
same => n,NoOP(${extension})
same => n,GotoIf($[“${extension}” = “”]?default)
same => n,Dial(sip/${extension},60,T)
same => n,Hangup()
same => n(default),Dial(sip/RAMAL_DEFAULT,60,T) ;Default route if the extension is not found in the database
same => n,Hangup()
Via SIP Redirect
We suggest reading this article before proceeding with ATI-EF via SIP Redirect: http://blog.sippulse.com/consulta-a-portabilidade-numerica-para-asterisk/
Sainte Call:
[from-internal]
exten => _X.,1,SipAddHeader(X-Extension: ${CALLERID(num)})
same => n,Dial(SIP/ati/${EXTEN})
[from-ati]
exten =>_X.,1,NoOP(${EXTEN})
Incoming Call:
[from-pstn]
exten => s,1,SipAddHeader(X-Get-Extension: )
same => n,Dial(SIP/ati/${EXTEN})
[from-ati]
exten _X.,1,NoOP(${EXTEN})
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