[Freeswitch-users] From domain

[Michael S Collins]

You have "prifile1" instead of profile1. Typo?
-MC

Sent from my iPhone

On Jul 24, 2008, at 8:17 AM, "Alex Gusak" <alex.gusak at gmail.com> wrote:

> 2008/7/24 Brian West <brian at freeswitch.org>:
>> And you have a gateway configured in FS ?  do you auth against this
>> domain/gateway?
>
> I configured gateway in sip_profiles. Gateway configured without
> registration. Full config:
>
> <profile name="prifile1" domain="domain.net">
>  <gateways>
>
>  <gateway name="gw1" ip="X.X.X.X">
>  <param name="caller-id-in-from" value="true"/>
>  <param name="realm" value="domain.net"/>
>  <param name="from-domain" value="domain.net"/>
>  <param name="username" value="1"/>
>  <param name="password" value="1"/>
>  <param name="register" value="false"/>
>  </gateway>
>
>  <gateway name="gw2" ip="Y.Y.Y.Y">
>  <param name="caller-id-in-from" value="true"/>
>  <param name="realm" value="domain.net"/>
>  <param name="from-domain" value="domain.net"/>
>  <param name="username" value="1"/>
>  <param name="password" value="1"/>
>  <param name="register" value="false"/>
>  </gateway>
>
>
>  </gateways>
>
>  <settings>
>    <param name="debug" value="1"/>
>    <param name="sip-trace" value="yes"/>
>    <param name="context" value="default"/>
>    <param name="rfc2833-pt" value="101"/>
>    <param name="sip-port" value="5060"/>
>    <param name="dialplan" value="XML"/>
>    <param name="dtmf-duration" value="100"/>
>    <param name="codec-prefs" value="$${global_codec_prefs}"/>
>    <param name="use-rtp-timer" value="true"/>
>    <param name="rtp-timer-name" value="soft"/>
>    <param name="rtp-ip" value="$${local_ip_v4}"/>
>    <param name="sip-ip" value="$${local_ip_v4}"/>
>    <param name="enable-timer" value="false"/>
>    <param name="enable-100rel" value="false"/>
>    <param name="apply-inbound-acl" value="local"/>
>    <param name="manage-presence" value="false"/>
>    <param name="supress-cng" value="true"/>
>    <!--param name="vad" value="both"/-->
>    <param name="inbound-codec-negotiation" value="greedy"/>
>    <!--param name="inbound-codec-negotiation" value="generous"/-->
>    <param name="rtp-timeout-sec" value="120"/>
>    <param name="rtp-hold-timeout-sec" value="600"/>
>    <param name="inbound-late-negotiation" value="true"/>
>    <param name="inbound-no-media" value="false"/>
>    <param name="accept-blind-reg" value="false"/>
>    <param name="nonce-ttl" value="60"/>
>    <param name="auth-calls" value="false"/>
>  </settings>
> </profile>
>
>
>>
>> If you're not using a gateway and you need to control the invite
>> domain you must have at least rev 9127 you can set variable
>> sip_invite_domain
>>
>
>
> In last snapshot its work, thanks :)
>
> <action application="bridge"
> data="{sip_invite_domain=domain.net}sofia/gateway/gw2/$1 at domain.net"/>
>
>
> (without sip_invite_domain in From: FreeSWITCH send IP address)
>
>
> -- 
> Alex Gusak
>
> _______________________________________________
> 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