[Freeswitch-users] 300 Multiple Choices - how to force FS to replace From field
Lyubo Popov
koki.roul at gmail.com
Thu May 4 19:20:27 MSD 2017
Hello Brian,
The authentication is DIGEST done via RADIUS. The Username is the same as
the Caller ID...or maybe I understood your question wrong..? The number
that shows in the from field is actually the SIP username created in the
system ( for routing, billing, radius AAA, etc. ) and it is as well the
callers number ( Caller ID). I use Raduis AAA to authenticate and account
the calls and the user you see 551000 is actually a username of voip
account created in the billing. That is why you see the incoming call from
that username (551000). This is what FS is using in the FROM field. Since
many of the accounts are created in format different from E164, often it is
necessary to rewrite the account number ( the caller number ) to E164 or
the termination will not accept the call. This is what I am trying to do
now, set a rewrite rule in the billing system to convert 551000 to
1140031556 and this is what it is returned to FS as you can see in the
packets the billing sends back..
Cheers,
L.Popov
On Wed, May 3, 2017 at 7:20 PM, Brian West <brian at freeswitch.org> wrote:
> Why are you using the from field for authentication?
>
> On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov <koki.roul at gmail.com> wrote:
>
>> 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 at 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 at 177.41.146.98:1048;transport=UDP>
>> To: <sip:556230951662 at 216.x.x.x:5080;transport=UDP>
>> From: "551000"<sip:551000 at 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 at 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 at 177.x.x.x>;tag=FeNXS71300N0c
>> To: <sip:556230951662 at 69.x.x.x:5060>
>> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b
>> CSeq: 106579790 INVITE
>> Contact: <sip:mod_sofia at 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 at 216.x.x.x>;party=c
>> alling;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 at 177.x.x.x>;tag=FeNXS71300N0c
>> To: <sip:556230951662 at 69.x.x.x:5060>
>> Contact: <sip:556230951662 at 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 at 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 at 216.x.x.x>;tag=gQepU2j7X9BKr
>> To: <sip:556230951662 at 162.x.x.x:5060>
>> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b
>> CSeq: 106579790 INVITE
>> Contact: <sip:mod_sofia at 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 at 216.x.x.x>;party=c
>> alling;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=${accountcode}"
>> />
>> <action application="set" data="call_direction=outbound" />
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="inherit_codec=true" />
>> <action application="set" data="call_timeout=20"/>
>> <action application="set" data="fail_on_single_reject=USER_BUSY"
>> />
>> <action application="set" data="origination_caller_id_na
>> me=${sip_req_user}"/>
>> <action application="set" data="origination_caller_id_nu
>> mber=${sip_from_user}"/>
>> <action application="set" data="execute_on_answer=sched_hangup
>> +${h323-credit-time} alloted_timeout" />
>> <action application="bridge" data="{sip_invite_from_uri=sip
>> :${sip_from_user}@${sip_network_ip}}sofia/internal/${destina
>> tion_number}@69.x.x.x:5060" />
>> <action application="hangup" data="${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
>>
>>
>>
>> _________________________________________________________________________
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>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
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