[Freeswitch-users] Custom Headers with Sip.js

Ítalo Rossi italorossib at gmail.com
Wed Mar 18 19:39:15 MSK 2015


You can set variables in your agent contact:

<agent name="1001 at default" type="callback"
contact="[origination_caller_id_name='Queue
Caller',call_timeout=10]user/1001 at 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 at 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 at gmail.com> wrote:
>
>> using mod_callcenter? try setting cc_export_vars
>>
>> On Wed, Mar 18, 2015 at 9:47 AM, Ali Jibran <ali.jibran44 at 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 at langfeld.co.uk> wrote:
>>>
>>>> Prefix the header with X-. Anything else is invalid.
>>>>
>>>> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at 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 at anonymous.invalid SIP/2.0
>>>>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>>>>    Max-Forwards: 70
>>>>>    To: <sip:900 at anonymous.invalid>
>>>>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>>>>    Call-ID: l5pieavpjguoset7ugvd
>>>>>    CSeq: 1458 INVITE
>>>>>   * a: foo*
>>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>>    Content-Type: application/sdp
>>>>>    Contact: <sip:8rgi365g at 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?
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
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>>>>> http://www.freeswitchsolutions.com
>>>>>
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>>>>>
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>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
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>>
>>
>>
>> --
>> Ítalo Rossi
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>



-- 
Ítalo Rossi
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