[Freeswitch-users] 300 Multiple Choices - how to force FS to replace From field
Lyubo Popov
lpopov at blasterphone.com
Mon May 15 06:23:13 MSD 2017
Hello Brian,
Sorry, did not see your response on May, 5th. Yes I have this in sofia
profile.
<param name="manual-redirect" value="true"/>
Is there anything else that I need to add or change in order to make FS to
use the returned FROM value by the SIP REDIRECT server?
Cheers,
L.Popov
On Fri, May 5, 2017 at 10:31 AM, Brian West <brian at freeswitch.org> wrote:
> are you setting manual-redirect in your Sofia profile?
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Handling+SIP+Redirect
>
> On Thu, May 4, 2017 at 10:20 AM, Lyubo Popov <koki.roul at gmail.com> wrote:
>
>> 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
>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>>
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>
>>> Allison prompts for FreeSWITCH:
>>>
>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>> *Skype:*briankwest
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
--
Atenciosamente,
============================
Lyubo Popov
CEO - BlasterPhone LLC
Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
iNum: +883 510001-354111
Website: http://www.blastervoip.com.br/
============================
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170514/94c91df0/attachment-0001.html
Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users
mailing list