We have sheme: +-----+ +----+ +-----+ | GW1 |-->| FS |-->| GW2 | +-----+ +----+ +-----+ <gateway name="GW1" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> <gateway name="GW2" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> When call from GW1 to GW2 FreeSWITCH recv INVITE from GW1: From: "Alex"<sip:alex_gusak@domain.net>;transport=UDP;tag=7552bb34 To: <sip:XXXXXX@domain.net>;transport=UDP and send INVITE to GW2: From: "Alex" <sip:alex_gusak@X.X.X.X>;tag=BK71NZ3yZvptK To: <sip:XXXXXX@domain.net> where X.X.X.X IP address FreeSWITCH. How to make that pass in the header From: the domain name domain.net? -- Alex Gusak
2008/7/24 Brian West <brian@freeswitch.org>:
Show us your bridge line .. how you call the gateway.
You want to see dialplan or channel vars? Dialplan looks like this: <context name="from_gw1"> <extension name="GW2"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="dialed_ext=$1"/> <action application="set" data="call_timeout=30"/> <action application="ring_ready"/> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/> <action application="set" data="continue_on_fail=NORMAL_CLEARING,NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/profile1/$1@domain.net"/> <action application="gentones" data="%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <action application="sleep" data="1300"/> <action application="gentones" data="%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <action application="sleep" data="1300"/> <action application="hangup"/> </condition> </extension> </context>
On Jul 24, 2008, at 8:46 AM, Alex Gusak wrote:
How to make that pass in the header From: the domain name domain.net?
Brian West sip:brian@freeswitch.org
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-- Alex Gusak
2008/7/24 Alex Gusak <alex.gusak@gmail.com>:
2008/7/24 Brian West <brian@freeswitch.org>:
Show us your bridge line .. how you call the gateway.
You want to see dialplan or channel vars?
Dialplan looks like this:
sorry, bridge in this dialplan look: <action application="bridge" data="sofia/profile1/$1@ip_address_gw2"/>
<context name="from_gw1"> <extension name="GW2"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="dialed_ext=$1"/> <action application="set" data="call_timeout=30"/> <action application="ring_ready"/> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/> <action application="set" data="continue_on_fail=NORMAL_CLEARING,NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/profile1/$1@domain.net"/> <action application="gentones" data="%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <action application="sleep" data="1300"/> <action application="gentones" data="%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <action application="sleep" data="1300"/> <action application="hangup"/> </condition> </extension> </context>
On Jul 24, 2008, at 8:46 AM, Alex Gusak wrote:
How to make that pass in the header From: the domain name domain.net?
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
-- Alex Gusak
-- Alex Gusak
And you have a gateway configured in FS ? do you auth against this domain/gateway? If you're not using a gateway and you need to control the invite domain you must have at least rev 9127 you can set variable sip_invite_domain /b On Jul 24, 2008, at 9:16 AM, Alex Gusak wrote:
sorry, bridge in this dialplan look:
<action application="bridge" data="sofia/profile1/$1@ip_address_gw2"/>
Brian West sip:brian@freeswitch.org
2008/7/24 Brian West <brian@freeswitch.org>:
And you have a gateway configured in FS ? do you auth against this domain/gateway?
I configured gateway in sip_profiles. Gateway configured without registration. Full config: <profile name="prifile1" domain="domain.net"> <gateways> <gateway name="gw1" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> <gateway name="gw2" ip="Y.Y.Y.Y"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="sip-trace" value="yes"/> <param name="context" value="default"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="enable-timer" value="false"/> <param name="enable-100rel" value="false"/> <param name="apply-inbound-acl" value="local"/> <param name="manage-presence" value="false"/> <param name="supress-cng" value="true"/> <!--param name="vad" value="both"/--> <param name="inbound-codec-negotiation" value="greedy"/> <!--param name="inbound-codec-negotiation" value="generous"/--> <param name="rtp-timeout-sec" value="120"/> <param name="rtp-hold-timeout-sec" value="600"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-no-media" value="false"/> <param name="accept-blind-reg" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> </settings> </profile>
If you're not using a gateway and you need to control the invite domain you must have at least rev 9127 you can set variable sip_invite_domain
In last snapshot its work, thanks :) <action application="bridge" data="{sip_invite_domain=domain.net}sofia/gateway/gw2/$1@domain.net"/> (without sip_invite_domain in From: FreeSWITCH send IP address) -- Alex Gusak
You have "prifile1" instead of profile1. Typo? -MC Sent from my iPhone On Jul 24, 2008, at 8:17 AM, "Alex Gusak" <alex.gusak@gmail.com> wrote:
2008/7/24 Brian West <brian@freeswitch.org>:
And you have a gateway configured in FS ? do you auth against this domain/gateway?
I configured gateway in sip_profiles. Gateway configured without registration. Full config:
<profile name="prifile1" domain="domain.net"> <gateways>
<gateway name="gw1" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway>
<gateway name="gw2" ip="Y.Y.Y.Y"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway>
</gateways>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="yes"/> <param name="context" value="default"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="enable-timer" value="false"/> <param name="enable-100rel" value="false"/> <param name="apply-inbound-acl" value="local"/> <param name="manage-presence" value="false"/> <param name="supress-cng" value="true"/> <!--param name="vad" value="both"/--> <param name="inbound-codec-negotiation" value="greedy"/> <!--param name="inbound-codec-negotiation" value="generous"/--> <param name="rtp-timeout-sec" value="120"/> <param name="rtp-hold-timeout-sec" value="600"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-no-media" value="false"/> <param name="accept-blind-reg" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> </settings> </profile>
If you're not using a gateway and you need to control the invite domain you must have at least rev 9127 you can set variable sip_invite_domain
In last snapshot its work, thanks :)
<action application="bridge" data="{sip_invite_domain=domain.net}sofia/gateway/gw2/$1@domain.net"/>
(without sip_invite_domain in From: FreeSWITCH send IP address)
-- Alex Gusak
_______________________________________________ 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
Remove the ip= attr as that doesn't do anything. Where ip= was documented on the wiki for use with a gateway? <param name="proxy" value="X.X.X.X"/> you would dial these gateways via sofia/gateway/gw1/destination (no @domain or anything) /b On Jul 24, 2008, at 10:17 AM, Alex Gusak wrote:
2008/7/24 Brian West <brian@freeswitch.org>:
And you have a gateway configured in FS ? do you auth against this domain/gateway?
I configured gateway in sip_profiles. Gateway configured without registration. Full config:
<profile name="prifile1" domain="domain.net"> <gateways>
<gateway name="gw1" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway>
<gateway name="gw2" ip="Y.Y.Y.Y"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway>
</gateways>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="yes"/> <param name="context" value="default"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="enable-timer" value="false"/> <param name="enable-100rel" value="false"/> <param name="apply-inbound-acl" value="local"/> <param name="manage-presence" value="false"/> <param name="supress-cng" value="true"/> <!--param name="vad" value="both"/--> <param name="inbound-codec-negotiation" value="greedy"/> <!--param name="inbound-codec-negotiation" value="generous"/--> <param name="rtp-timeout-sec" value="120"/> <param name="rtp-hold-timeout-sec" value="600"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-no-media" value="false"/> <param name="accept-blind-reg" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> </settings> </profile>
If you're not using a gateway and you need to control the invite domain you must have at least rev 9127 you can set variable sip_invite_domain
In last snapshot its work, thanks :)
<action application="bridge" data="{sip_invite_domain=domain.net}sofia/gateway/gw2/$1@domain.net"/>
(without sip_invite_domain in From: FreeSWITCH send IP address)
-- Alex Gusak
_______________________________________________ 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 sip:brian@freeswitch.org
2008/7/24 Brian West <brian@freeswitch.org>:
Remove the ip= attr as that doesn't do anything. Where ip= was documented on the wiki for use with a gateway?
You're right, I mix this with <user/> :)
<param name="proxy" value="X.X.X.X"/>
you would dial these gateways via sofia/gateway/gw1/destination (no @domain or anything)
Thanks! -- Alex Gusak
participants (3)
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Alex Gusak
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Brian West
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Michael S Collins