Hello, So I found this page which shows how to remove custom SIP headers by using the unset application and passing in the custom header name prefixed by sip_h_[HEADER]. Is there a configuration setting which can be set to disallow custom headers being copied when bridging calls? The scenario is as following; 1) Each call is going through a Kamailio SIP Proxy first, which adds some custom headers. 2) Kamailio routes the INVITE to one of our FS SBC's. 3) FS bridges the call to the carrier. When FS creates a new call leg towards the carrier, I would expect it to NOT copy the custom headers added by Kamailio, but it does. This is my dialplan: <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="privacy" data="full"/> <action application="set" data="sip_h_Privacy=id"/> <action application="set" data="privacy=yes"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Did I configure something wrong, or is this just normal behavior?
Hi Grant, IMHO there is no way dont copy the sip-x header to the new call-leg. The simple way to resolve this, is to remove your kamailio sip-x headers after you read it and before you create the b-leg. http://wiki.freeswitch.org/wiki/Strip_SIP_Headers 2013/8/26 Grant Bagdasarian <GB@cm.nl>
Hello,****
** **
So I found this page which shows how to remove custom SIP headers by using the unset application and passing in the custom header name prefixed by sip_h_[HEADER].****
Is there a configuration setting which can be set to disallow custom headers being copied when bridging calls? ****
** **
The scenario is as following; ****
**1) **Each call is going through a Kamailio SIP Proxy first, which adds some custom headers.****
**2) **Kamailio routes the INVITE to one of our FS SBC’s.****
**3) **FS bridges the call to the carrier.****
** **
When FS creates a new call leg towards the carrier, I would expect it to NOT copy the custom headers added by Kamailio, but it does. ****
** **
This is my dialplan:****
** **
<extension name="internal_to_external">****
<condition field="${sip_req_host}" expression="^10\.0\.0\.1$">****
<action application="set" data="hangup_after_bridge=false"/>****
<action application="set" data="continue_on_fail=true"/>****
<action application="privacy" data="full"/>****
<action application="set" data="sip_h_Privacy=id"/>****
<action application="set" data="privacy=yes"/>****
<action application="bridge" data="sofia/external/${sip_req_uri}"/> ****
</condition>****
</extension>****
** **
Did I configure something wrong, or is this just normal behavior?****
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Hello, Yeah, that's the page I was talking about. I also found this: http://wiki.freeswitch.org/wiki/Variable_sip_copy_custom_headers This action works. It strips all the custom headers. It would have been nice if I could configure this in a sip_profile. <action application="set" data="sip_copy_custom_headers=false" /> Is the sip_copy_custom_headers variable a channel variable or can it also be set in a sip_profile? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Karsten Horsmann Sent: Monday, August 26, 2013 9:54 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Remove custom SIP headers Hi Grant, IMHO there is no way dont copy the sip-x header to the new call-leg. The simple way to resolve this, is to remove your kamailio sip-x headers after you read it and before you create the b-leg. http://wiki.freeswitch.org/wiki/Strip_SIP_Headers 2013/8/26 Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> Hello, So I found this page which shows how to remove custom SIP headers by using the unset application and passing in the custom header name prefixed by sip_h_[HEADER]. Is there a configuration setting which can be set to disallow custom headers being copied when bridging calls? The scenario is as following; 1) Each call is going through a Kamailio SIP Proxy first, which adds some custom headers. 2) Kamailio routes the INVITE to one of our FS SBC's. 3) FS bridges the call to the carrier. When FS creates a new call leg towards the carrier, I would expect it to NOT copy the custom headers added by Kamailio, but it does. This is my dialplan: <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="privacy" data="full"/> <action application="set" data="sip_h_Privacy=id"/> <action application="set" data="privacy=yes"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Did I configure something wrong, or is this just normal behavior? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 -- Mit freundlichen Grüßen *Karsten Horsmann*
participants (2)
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Grant Bagdasarian
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Karsten Horsmann