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?
That would have worked if the variable you put into the message was X-a instead of a
On Mar 18, 2015, at 6:51 AM, 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?
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?
_________________________________________________________________________ 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
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 <javascript:_e(%7B%7D,'cvml','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?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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
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?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
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 <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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?
_________________________________________________________________________ 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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
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?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
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 <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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
The “export” app can. Bote From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Jibran Sent: Wednesday, 18 March, 2015 12:43 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Custom Headers with Sip.js Yeah but I can't pass a variable from a leg to b can I?
Can you explain in detail Please? I need to export a header from A leg to B leg....but in the call Center app. Like the variable should be exported to whoever picks up the call. On Wednesday, March 18, 2015, Bote Man <bote_radio@botecomm.com> wrote:
The “export” app can.
Bote
*From:* freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');>] *On Behalf Of *Ali Jibran *Sent:* Wednesday, 18 March, 2015 12:43 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Custom Headers with Sip.js
Yeah but I can't pass a variable from a leg to b can I?
Sadly thats no longer true, Crazy right? On Wed, Mar 18, 2015 at 7:28 AM, 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?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
participants (6)
-
Ali Jibran
-
Ben Langfeld
-
Bote Man
-
Brian West
-
Michael Jerris
-
Ítalo Rossi