[Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip

Bryan Lemon bryan at bryanlemon.com
Wed Aug 17 19:10:17 MSD 2011


<profile name="external">
  <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"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${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="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <!--<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="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <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="tls" value="$${external_ssl_enable}"/>
    <param name="tls-bind-params" value="transport=tls"/>
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>


Thank you,
Bryan Lemon
(302) 648-2747



On Tue, Aug 16, 2011 at 13:03, Brian West <brian at freeswitch.org> wrote:

> Bryan,
>        Can you provide the sofia profile xml?
>
> /b
>
> On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
>
> >> From what I am seeing, freeswitch is not honoring the ext-*-ip variables
> in
> > the invite messages. Using the following command entered on
> > fs_cli: originate
> >
> {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
> > &javascript(somejavascript.js), the invite message is below. Shouldn't
> the
> > instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*?
> This
> > is causing the rtp packets to be sent to the incorrect location, and
> > resulting in 1-way audio.
> >
> >
> > send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
> >
> ------------------------------------------------------------------------
> >   INVITE sip:13044150838<
> https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.*
> > SIP/2.0
> >   Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD
> >   Max-Forwards: 69
> >   From: "SomeName" <sip:bryan_bryanlemon_com at 209.216.
> > *.*;transport=udp>;tag=HpU27XSQHmX1g
> >   To: <sip:15555551212 at 209.216.*.*>
> >   Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
> >   CSeq: 16401016 INVITE
> >   Contact: <sip:gw+didforsale at 204.111.
> *.*:5080;transport=udp;gw=didforsale>
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11
> 14-15-26
> > -0500
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY
> >   Supported: timer, precondition, path, replaces
> >   Allow-Events: talk, hold, refer
> >   Content-Type: application/sdp
> >   Content-Disposition: session
> >   Content-Length: 203
> >   X-FS-Support: update_display
> >   Remote-Party-ID: "SomeName" <sip:15555551212 at 10.0.10.144
> >> ;party=calling;screen=yes;privacy=off
> >
> >   v=0
> >   o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144
> >   s=FreeSWITCH
> >   c=IN IP4 10.0.10.144
> >   t=0 0
> >   m=audio 32712 RTP/AVP 8 0 3 101 13
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-16
> >   a=ptime:20
> >
> >
> >
> >
> > sofia status profile internal
> >
> =================================================================================================
> > Name              internal
> > Domain Name       N/A
> > Auto-NAT          true
> > DBName            sofia_reg_internal
> > Pres Hosts        10.0.10.144,10.0.10.144
> > Dialplan          XML
> > Context           public
> > Challenge Realm   auto_from
> > RTP-IP            10.0.10.144
> > Ext-RTP-IP        204.111.*.*
> > SIP-IP            10.0.10.144
> > Ext-SIP-IP        204.111.*.*
> > URL               sip:mod_sofia at 10.0.10.144:5060
> > BIND-URL          sip:mod_sofia at 10.0.10.144:5060
> >
> >
> > freeswitch at internal> sofia status profile external
> >
> =================================================================================================
> > Name              external
> > Domain Name       N/A
> > Auto-NAT          true
> > DBName            sofia_reg_external
> > Pres Hosts
> > Dialplan          XML
> > Context           public
> > Challenge Realm   auto_to
> > RTP-IP            10.0.10.144
> > Ext-RTP-IP        204.111.*.*
> > SIP-IP            10.0.10.144
> > Ext-SIP-IP        204.111.*.*
> > URL               sip:mod_sofia at 10.0.10.144:5080
> > BIND-URL          sip:mod_sofia at 10.0.10.144:5080
> >
> >
> > Thank you,
> > Bryan Lemon
> > (302) 648-2747 <
> https://www.google.com/voice/m/caller?number=+13026482747>
> >
> > 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
>
>
>
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