Hello everyone,
I would like to express my thanks in advance to anyone who may be able to help me with some insides.
I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process.
1. Sending call to FS with Zoiper, destination 556230951662
INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK-d8754z-038f1c7d251308c2-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:551000@177.41.146.98:1048;transport=UDP>
To: <sip:556230951662@216.x.x.x:5080;transport=UDP>
From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20
Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Allow-Events: presence, kpml
Content-Length: 232
v=0
o=Zoiper_user 0 0 IN IP4 177.x.x.x
s=Zoiper_session
c=IN IP4 177.x.x.x
t=0 0
m=audio 8000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
2. FS sending INVITE to SIP Redirect server
INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa
Max-Forwards: 69
From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c
To: <sip:556230951662@69.x.x.x:5060>
Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b
CSeq: 106579790 INVITE
Contact: <sip:mod_sofia@216.x.x.x:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 397
X-FS-Support: update_display,send_info
Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x
2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x
s=FreeSWITCH
c=IN IP4 216.x.x.x
t=0 0
m=audio 22476 RTP/AVP 8 0 18 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=audio 22476 RTP/AVP 4 101 13
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556)
SIP/2.0 300 Multiple Choices
Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa
From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c
To: <sip:556230951662@69.x.x.x:5060>
Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00
Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b
CSeq: 106579790 INVITE
Max-Forwards: 69
Content-Length: 0
Server: SIP Redirect Server
4. FS will send the call to the Termination IP WITHOUT changing the FROM field
INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp
Max-Forwards: 68
From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr
To: <sip:556230951662@162.x.x.x:5060>
Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b
CSeq: 106579790 INVITE
Contact: <sip:mod_sofia@216.x.x.x:5060>
2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 397
X-FS-Support: update_display,send_info
Remote-Party-ID: "551000" <sip:551000@216.x.x.x>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x
s=FreeSWITCH
c=IN IP4 216.x.x.x
t=0 0
m=audio 20696 RTP/AVP 8 0 18 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=audio 20696 RTP/AVP 4 101 13
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
This is the dialplan I use to send calls to SIP redirect server
<include>
<extension name="rejections">
<condition field="${radius_auth_result}" expression="2">
<action application="hangup" data="CALL_REJECTED"/>
</condition>
</extension>
<extension name="timedouts">
<condition field="${radius_auth_result}" expression="1">
<action application="hangup" data="SWITCH_CONGESTION"/>
</condition>
</extension>
<extension name="SIP Redirect Server">
<condition field="${radius_auth_result}" expression="0"/>
<condition field="destination_number" expression="^(.+)$">
<!--<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/internal/${destination_number}@69.x.x.x:5060" />
<action application="hangup" data="${bridge_hangup_cause}"/>
</condition>
</extension>
</include
Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated!
Best regards,
L. Popov