Press 1 config call transfer to Asterisk call queue (works)

Moderator: areski

Press 1 config call transfer to Asterisk call queue (works)

Postby fintech » Thu May 02, 2013 3:29 pm

We are going to test press 1 today. We have a newfies dialer and elastix(asterisk) connected via sip trunk. If somebody presses 1 then the call goes through the SIP trunk to elastix via a SIP URI

Newfies = 192.168.1.4
Elastix = 192.168.1.2


Newfies SIP trunk config to elastix:

<include>
<gateway name="192.168.1.2">
<!--/// account username *required* ///-->
<!--param name="Intelepeer" value="Intelepeer"/>-->
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<!--<param name="realm" value="asterlink.com"/>-->
<!--/// username to use in from: *optional* same as username, if blank ///-->
<!--param name="from-user" value="your username provided by carrier"/>-->
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
<!--param name="from-domain" value=""/-->
<!--/// account password *required* ///-->
<!--/// param name="password" value="your password supplied by carrier"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<!--<param name="proxy" value="asterlink.com"/>-->
<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
<!--<param name="register-proxy" value="mysbc.com"/>-->
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<!--<param name="expire-seconds" value="60"/>-->
<!--/// do not register ///-->
<param name="register" value="false"/>
<!-- which transport to use for register -->
<!--<param name="register-transport" value="udp"/>-->
<!--How many seconds before a retry when a failure or timeout occurs -->
<!--<param name="retry-seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
<!--<param name="caller-id-in-from" value="false"/>-->
<!--extra sip params to send in the contact-->
<!--<param name="contact-params" value="tport=tcp"/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
</gateway>
</include>

Elastix SIP trunk to Newfies:

allow=all
type=friend
host=192.168.1.4
dtmfmode=rfc2833
context=from-internal
qualify=no
insecure=very


The Elastix call queue pilot number is 270

In newfies survey select call transfer and put in 270@192.168.1.2 in the SIP URI box. This works and I have tested it. We are going to go into test production with config today.
fintech
 
Posts: 18
Joined: Mon Feb 18, 2013 12:23 am
Location: Newport Beach, CA

Re: Press 1 config call transfer to Asterisk call queue (wor

Postby pmurphy » Wed May 08, 2013 3:25 pm

I have been testing exactly same setup with elastix. When I created the transfer section it seems to transfer without pressing the 1 as would be required in an IVR option.
I created the transfer to a regular extension on elastix and it did the transfer and allowed me to leave voice mail.

Have you tested the transfer further?

Did it require you pressing the digit?

My concept was to use the newfies platform to deliver messages to customer, then play message option to talk to live agent, so it wouldn't transfer calls answered by an answering machine only the live customers that press an option.

Maybe I am missing something. Help appreciated if you have some to offer.
pmurphy
 
Posts: 19
Joined: Mon Apr 01, 2013 11:41 pm

Re: Press 1 config call transfer to Asterisk call queue (wor

Postby areski » Wed May 08, 2013 3:31 pm

Hi,

In version 2.1, we introduced Call transfer, which allow you to transfer a Call to any SIP point or to an other phone number :
http://www.newfies-dialer.org/press-1-l ... -dnc-list/

You can easily play a greeting message on the survey and then after the user press a DTMF for instance 1, go to an other section of the survey to perform the Call Transfer.

Yours,
/Areski
areski
Site Admin
 
Posts: 302
Joined: Tue Oct 18, 2011 5:28 pm

Re: Press 1 config call transfer to Asterisk call queue (wor

Postby pmurphy » Wed May 08, 2013 7:54 pm

I think I have it now.
Survey:
1. Play message
2. Capture digits (branch to call transfer with press 1)
3. Call tranfer (using sup uri formed as sofia/gateway/"name of gatway to elastix"/extension or que, so mine looks like sofia/gateway/elastix/202 to transfer calls to extension 202

I like the application. Would be happy to pay for a book on configurations to save some time. How about a training class on the web?
pmurphy
 
Posts: 19
Joined: Mon Apr 01, 2013 11:41 pm

Re: Press 1 config call transfer to Asterisk call queue (wor

Postby areski » Wed May 08, 2013 10:35 pm

We do provide training for this, this is included in our services:
http://www.newfies-dialer.org/pricing/

Yours,
/Areski
areski
Site Admin
 
Posts: 302
Joined: Tue Oct 18, 2011 5:28 pm


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