[Freeswitch-users] Profile misunderstanding

Sergey Safarov s.safarov at gmail.com
Fri Jan 13 12:30:46 MSK 2017


First required to fix
Ext-SIP-IP          <EXT_ADDR>
To do it edit "default" profile setting and set ext-sip-ip and ext-rtp-ip
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/sip_profiles/internal.xml#284-285>
to
ip address values.


пт, 13 янв. 2017 г. в 12:22, Igor Olhovskiy <igorolhovskiy at gmail.com>:

> Hi!
> I have profile named default:
> sofia status profile default
>
> =================================================================================================
> Name                default
> Domain Name         N/A
> Auto-NAT            false
> DBName              sofia_reg_default
> Pres Hosts          172.31.1.100,172.31.1.100
> Dialplan            XML
> Context             default
> Challenge Realm     auto_from
> RTP-IP              172.31.1.100
> Ext-RTP-IP          <EXT_ADDR>
> SIP-IP              172.31.1.100
> Ext-SIP-IP          <EXT_ADDR>
> URL                 sip:mod_sofia@<EXT_ADDR>:5060
> BIND-URL            sip:mod_sofia@
> <EXT_ADDR>:5060;maddr=172.31.1.100;transport=udp,tcp
> HOLD-MUSIC          local_stream://moh
> OUTBOUND-PROXY      N/A
> CODECS IN           PCMA,PCMU,G729
> CODECS OUT          PCMA,PCMU,G729
> TEL-EVENT           101
> DTMF-MODE           rfc2833
> CNG                 13
> SESSION-TO          0
> MAX-DIALOG          0
> NOMEDIA             false
> LATE-NEG            false
> PROXY-MEDIA         false
> ZRTP-PASSTHRU       false
> AGGRESSIVENAT       true
>
>
> Also:
>
> sip-ip [172.31.1.100]
> sip-port [5060]
> rtp-ip [172.31.1.100]
> dialplan [XML]
> user-agent-string [ASTPP]
> debug [0]
> sip-trace [no]
> tls [false]
> inbound-reg-force-matching-username [true]
> disable-transcoding [true]
> all-reg-options-ping [false]
> unregister-on-options-fail [true]
> log-auth-failures [true]
> status [0]
> inbound-bypass-media [false]
> inbound-proxy-media [false]
> disable-transfer [true]
> enable-100rel [false]
> rtp-timeout-sec [400]
> dtmf-duration [2000]
> manual-redirect [true]
> aggressive-nat-detection [true]
> enable-timer [false]
> minimum-session-expires [120]
> session-timeout-pt [1800]
> auth-calls [true]
> apply-inbound-acl [default]
> inbound-codec-prefs [PCMA,PCMU,G729]
> outbound-codec-prefs [PCMA,PCMU,G729]
> inbound-late-negotiation [false]
> sip-capture [no]
> forward-unsolicited-mwi-notify [false]
> context [default]
> rfc2833-pt [101]
> rtp-timer-name [soft]
> hold-music [local_stream://moh]
> manage-presence [true]
> presence-hosts [172.31.1.100,172.31.1.100]
> presence-privacy [false]
> inbound-codec-negotiation [generous]
> auth-all-packets [false]
> ext-rtp-ip [<EXT_ADDR>]
> ext-sip-ip [<EXT_ADDR>]
> rtp-hold-timeout-sec [1800]
> force-register-domain [172.31.1.100]
> force-subscription-domain [172.31.1.100]
> force-register-db-domain [172.31.1.100]
> challenge-realm [auto_from]
> nonce-ttl [60]
> pass-callee-id [false]
> rtcp-audio-interval-msec [5000]
> local-network-acl [localnet.auto]
> NDLB-force-rport [true]
>
> But problem is, when client tries to connect to FS, it still answers to
> port 5060, not looking on NDLB-force-rport option. As well, as private
> addresses shown across packets, not using ext-sip and ext-rtp options.
>
> ------------------------------------------------------------------------
> recv 378 bytes from udp/[136.169.20.219]:*3589* at 19:54:34.619718:
> ------------------------------------------------------------------------
> REGISTER sip:172.31.1.100:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305
> From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
> To: <sip:6399081327 at 172.31.1.100:5060>
> Call-ID: 1500202051 at 192.168.0.101
> CSeq: 197 REGISTER
> Contact: <sip:6399081327 at 192.168.0.101:5060>;expires=5
> Max-Forwards: 30
> User-Agent: dble
> Expires: 5
> Content-Length: 0
>
> ------------------------------------------------------------------------
> send 571 bytes to udp/[136.169.20.219]:*5060* at 19:54:34.619864:
> ------------------------------------------------------------------------
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.101:5060
> ;branch=z9hG4bK322184305;received=136.169.20.219
> From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
> To: <sip:6399081327 at 172.31.1.100:5060>;tag=jD7SQFyXcK1gr
> Call-ID: 1500202051 at 192.168.0.101
> CSeq: 197 REGISTER
> User-Agent: ASTPP
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> NOTIFY, PUBLISH, SUBSCRIBE
> Supported: path, replaces
> WWW-Authenticate: Digest realm="172.31.1.100",
> nonce="7d83af94-2a69-40cc-a9c3-1659945a34b2", algorithm=MD5, qop="auth"
> Content-Length: 0
>
>
> Can you please point what I’m missing?
> Thanks!
>
> Regards, Igor
> _________________________________________________________________________
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170113/ca9c7eae/attachment-0001.html 


Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list