[Freeswitch-users] Custom Headers with Sip.js

[Ali Jibran]

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
> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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?
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>>
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>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
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>>
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>
>
>
> --
> Ítalo Rossi
>
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