In your dialplan (30_routing.xml) set sip_redirect_context=to_anything. Then when your Jerasoft will send you back the redirect you will be able to catch the call in the context you defined above and do the bridge using sip_redirct_contact_0 to your internal profile to find the local user. Srigo Sent from my iPhone
On 20 Jun 2017, at 21:01, Jospeh Waite <joelists@tm.net.uk> wrote:
Hi Guys.
Final question of the day. I promise!!
I have 2 sofia profiles, one for IP authenticated calls and one for Sip Registrations. There both on port 5060 on 2 different IP addresses. This is basically a Wholesale/retail billing/switching platform. We have a JeraSoft VCS radius based billing and Sip Redirect server.
Everything works ok, except for inbound calls from DID provider destined for a registered extension.
Call comes in on the IP profile, which then sends a request to the JeraSoft SIP Redirect server, which replies with a 300 multiple choices as follows.
SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP {ip of FS IP profile};rport;branch=z9hG4bKSKFBFy6K81U9H From: "07966677711" <sip:07966677711@{ip of did provider}>;tag=S6Xmmj7meS2Fr To: <sip:443307881011@{ip of jerasoft redirect}:5060> Contact: <sip:joehouse@{ip of FS IP profile}>;q=1.00 Call-ID: aed22c59-d082-1235-baa2-363165383663 CSeq: 108654212 INVITE Max-Forwards: 67 Content-Length: 0 Server: JeraSoft VCS SIP Redirect Server
joehouse is the username of a registered user, registered to the reg profile.
Now FS instead of connecting the call to the registered user it sends a new request to the Jerasoft redirect.
How would I get it to route the call to the registered user?
public/30_routing.xml <?xml version="1.0"?> <include> <extension name="JeraSoft VCS Routing"> <!-- <condition field="${radius_auth_result}" expression="0"/>
<condition field="${h323-redirect-number}" expression="^(.+)$" break="never"> <action application="set" data="destination_number=$1" /> </condition> -->
<condition field="destination_number" expression="^(.+)$"> <!--<action application="info"/>--> <action application="export" data="nolocal:h323-call-origin=originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_name=${sip_req_user}"/> <action application="set" data="origination_caller_id_number=${sip_from_user}”/> <!-- <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> --> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@{Jerasoft SIP Redirect IP}:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include>
public.xml <profile name="external"> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <!-- <alias name="outbound"/> <alias name="nat"/> --> </aliases>
<domains> <domain name="all" alias="false" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. --> <!-- <param name="shutdown-on-fail" value="true"/> --> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <!-- RFC 5626 : Send reg-id and sip.instance --> <!--<param name="enable-rfc-5626" value="true"/> --> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <!--<param name="enable-100rel" value="true"/>--> <!--<param name="disable-srv503" value="true"/>--> <!-- This could be set to "passive" --> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="passive"/>
<!-- used to share presence info across sofia profiles manage-presence needs to be set to passive on this profile if you want it to behave as if it were the internal profile for presence. --> <!-- Name of the db to use for this profile --> <!--<param name="dbname" value="share_presence"/>--> <!--<param name="presence-hosts" value="$${domain}"/>--> <!--<param name="force-register-domain" value="$${domain}"/>--> <!--all inbound reg will stored in the db using this domain --> <!--<param name="force-register-db-domain" value="$${domain}"/>--> <!-- ************************************************* -->
<!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) --> <!-- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS! --> <param name="rtp-ip" value=“{IP address}"/> <param name="sip-ip" value=“{ip address}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <!--<param name="enable-3pcc" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="$${external_ssl_enable}"/> <!-- Set to true to not bind on the normal sip-port but only on the TLS port --> <param name="tls-only" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) --> <param name="tls-sip-port" value="5061"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) --> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files --> <param name="tls-passphrase" value=""/> <!-- Verify the date on TLS certificates --> <param name="tls-verify-date" value="true"/> <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate --> <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe --> <param name="tls-verify-policy" value="none"/> <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none --> <param name="tls-verify-depth" value="2"/> <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe --> <param name="tls-verify-in-subjects" value=""/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 --> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile>
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