Yeah but I can't pass a variable from a leg to b can I? 

On Wednesday, March 18, 2015, Ítalo Rossi <italorossib@gmail.com> wrote:
You can set variables in your agent contact:

<agent name="1001@default" type="callback" contact="[origination_caller_id_name='Queue Caller',call_timeout=10]user/1001@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />

And probably you can set variables to be expanded, in mod_callcenter.c I see

dialstr = switch_channel_expand_variables(member_channel, h->originate_string);

h->originate_string is the contact of the agent

So it'll probably work.

NOT TESTED!

On Wed, Mar 18, 2015 at 12:10 PM, Ali Jibran <ali.jibran44@gmail.com> wrote:
Didn't work :/

So there's no way I can export data to mod_callcenter? 


On Wednesday, March 18, 2015, Ítalo Rossi <italorossib@gmail.com> wrote:
using mod_callcenter? try setting cc_export_vars

On Wed, Mar 18, 2015 at 9:47 AM, Ali Jibran <ali.jibran44@gmail.com> wrote:
Ok yeah :) thank you that worked.

One more thing. I can access the Header now in my dialplan. But I pass the call into call Center.
Now how do I pass the header into call Center so that it pops up to which ever agent picked it up.

Like
Web->fs->call center->agent

I can get the header to Freeswitch but it's not showing after the call is bridged to agent.
I tried sip_copy_custom_headers=true but it didn't work

On Wednesday, March 18, 2015, Ben Langfeld <ben@langfeld.co.uk> wrote:
Prefix the header with X-. Anything else is invalid.

On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44@gmail.com> wrote:
Hi all.
I make a call to FS callcenter using webRTC. Now I would like to add some additional data to the header.
That I did by using extraheaders. Now all that is good. I can see the custom header in the sip trace on freeswitch.

For e.g in sip.js I add:
var options = {
  extraHeaders: [ 'a: foo' ]
};


on FS i get:
 INVITE sip:900@anonymous.invalid SIP/2.0
   Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
   Max-Forwards: 70
   To: <sip:900@anonymous.invalid>
   From: <sip:anonymous.hfsqa4@anonymous.invalid>;tag=depmif1fo5
   Call-ID: l5pieavpjguoset7ugvd
   CSeq: 1458 INVITE
   a: foo
   Contact: <sip:8rgi365g@9jscv0q24s4m.invalid;transport=ws;ob>
   Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
   Content-Type: application/sdp
   Contact: <sip:8rgi365g@9jscv0q24s4m.invalid;transport=ws;ob>
   Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
   Content-Type: application/sdp
   Supported: outbound
   User-Agent: SIP.js/0.5.0
   Content-Length: 3636

How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't work. Help please?

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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--
Ítalo Rossi

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Ítalo Rossi