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

Wed Aug 17 20:56:50 MSD 2011

if you know the exact value of the ip use that in place of the word
"auto-nat" which is specifically designed for nat-pnp enabled routers.

On Tue, Aug 16, 2011 at 12:07 PM, Bryan Lemon <bryan at bryanlemon.com> wrote:
> <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
>>
>>
>>
>> FreeSWITCH-users mailing list
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>
>
>
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>

-- 
Anthony Minessale II

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