<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@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@209.216.
> *.*;transport=udp>;tag=HpU27XSQHmX1g
>   To: <sip:15555551212@209.216.*.*>
>   Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
>   CSeq: 16401016 INVITE
>   Contact: <sip:gw+didforsale@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@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@10.0.10.144:5060
> BIND-URL          sip:mod_sofia@10.0.10.144:5060
>
>
> freeswitch@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@10.0.10.144:5080
> BIND-URL          sip:mod_sofia@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@lists.freeswitch.org
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