[Freeswitch-users] Phone registration error
Jair Santos
jairds at shaw.ca
Tue Jul 15 11:07:38 PDT 2008
Hi ,
I am strugling to make two phones to make calls. I 'll appreciate
information about the configs that I am listing below.
1s phone - Is an internat phone(behind NAT). This one was working before. It
was able to make calls to PSTN and to another extension.
The config is like
Sip server address: 192.168.1.117
Sip user id : 1000
2nd phone - It is external the of the Network.
The config is like :
Username : 1001
Domain 24.67.78.200:5060
Both are registering in FS. I cannot call between them.
When I try to make PSTN calls using external phone I get .
2008-07-15 10:45:34 [NOTICE] switch_channel.c:533 switch_channel_set_name()
New Channel sofia/internal2/1001 at 24.67.78.200:5060
[58e39d4a-e78f-4c96-8f43-3e969704d9e1]
2008-07-15 10:45:34 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing
FreeSwitch->12508606894 at public
2008-07-15 10:45:34 [ERR] mod_sofia.c:1713 sofia_outgoing_channel() Invalid
Gateway
2008-07-15 10:45:34 [NOTICE] mod_sofia.c:1904 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-07-15 10:45:34 [ERR] switch_ivr_originate.c:823 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[INVALID_NUMBER_FORMAT]
2008-07-15 10:45:34 [INFO] mod_dptools.c:1797 audio_bridge_function()
Originate Failed. Cause: INVALID_NUMBER_FORMAT
2008-07-15 10:45:34 [NOTICE] mod_dptools.c:1824 audio_bridge_function()
Hangup sofia/internal2/1001 at 24.67.78.200:5060 [CS_EXECUTE]
[INVALID_NUMBER_FORMAT]
2008-07-15 10:45:34 [NOTICE] switch_core_session.c:753
switch_core_session_thread() Session 17
(sofia/internal2/1001 at 24.67.78.200:5060) Ended
2008-07-15 10:45:34 [NOTICE] switch_core_session.c:755
switch_core_session_thread() Close Channel
sofia/internal2/1001 at 24.67.78.200:5060 [CS_HANGUP]
Please check the settings that I have.
On /conf/dialplan/extensions
teste.xml
<extension name="teste">
<condition field="destination_number" expression="^(\d+)$">
<!--- The % behind the username tells FS to lookup the user in it's
local sip_registration database -->
<action application="bridge" data="sofia/gateway/inphonex/$0"/>
</condition>
</extension>
On conf/sip_profiles
internal2.xml
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="internal2">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<aliases>
<alias name="internal2"/>
</aliases>
<domains>
<domain name="$${domain}" parse="true"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="24.67.78.200"/>
<param name="ext-sip-ip" value="24.67.78.200"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>
doublenat.xml
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="doublenat">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<aliases>
<alias name="doublenat"/>
</aliases>
<domains>
<domain name="$${domain}" parse="true"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="force-register-domain" value="$${domain}"/>
<param name="apply-nat-acl" value="rfc1918"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5090"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>
On /conf/directory/default
inphonex.xml
<include>
<user id="inphonex">
<gateways>
<gateway name="inphonex">
<param name="username" value="3462101"/>
<param name="realm" value="sip.varphonex.com"/>
<param name="password" value="606545"/>
<param name="proxy" value="sip.varphonex.com"/>
<param name="register" value="true"/>
<param name="expire-seconds" value="3600"/>
</gateway>
</gateways>
<params>
<param name="password" value="606545"/>
</params>
</user>
</include>
Thank you
Jair santos
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