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Thanks for all your input guys !!!<br>
Now thiz newbie can pass the traffic from A to B SIP server <span
class="moz-smiley-s1"><span> :-) </span></span><br>
<br>
regards,<br>
<br>
~pieter~<br>
<br>
Jonas Gauffin wrote:
<blockquote
cite="midd021d3be0709200855v28198c7dj2eecc2620dd8544f@mail.gmail.com"
type="cite">
<pre wrap="">Your freeswitch.xml is invalid.
It should be "<!--#set "sip_profile=1.2.3.4" -->"
and not "#set "sip_profile=1.2.3.4""
I made the same mistake myself, when I first configured freeswitch.
The profile name should also match the one in sofia.conf.xml.
The profiles is used to be able to configure multiple sites in the
same switch. To make it easy, use the domain name as the profile name.
On 9/20/07, Michael Jerris <a class="moz-txt-link-rfc2396E" href="mailto:mike@jerris.com"><mike@jerris.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap=""><a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a>
that "//" looks not right. But would need to see the log above the snipet
you put in to see what is going on for sure.
On 9/20/07, Brian West <a class="moz-txt-link-rfc2396E" href="mailto:brian.west@mac.com"><brian.west@mac.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">
Not Acceptable Here would indicate aa codec isssue. As would
</pre>
</blockquote>
<pre wrap="">INCOMPATIBLE_DESTINATION as the hangup cause. Can you verify the codec
configs on each end?
</pre>
<blockquote type="cite">
<pre wrap="">
/b
On Sep 20, 2007, at 12:19 AM, Pieter Eduard wrote:
Hi,
I just installed Freeswitch and my intention to use fs to route calls
</pre>
</blockquote>
<pre wrap="">between gateways but am still stuck with the proper configuration.
</pre>
<blockquote type="cite">
<pre wrap="">The simple diagram is like this :
subscriber A------> SIP Server A ------>Freeswitch----->SIP Server B----->
</pre>
</blockquote>
<pre wrap="">subscriber B
</pre>
<blockquote type="cite">
<pre wrap="">voip prefix ip a.b.c.d ip 1.2.3.4
</pre>
</blockquote>
<pre wrap=""> ip w.x.y.z voip prefix
</pre>
<blockquote type="cite">
<pre wrap="">777888x
</pre>
</blockquote>
<pre wrap=""> 999111x
</pre>
<blockquote type="cite">
<pre wrap="">I tried to call subscriber B from subscriber A but getting this error :
2007-09-20 10:40:24 [NOTICE] sofia.c:1171 sofia_handle_sip_i_state()
</pre>
</blockquote>
<pre wrap="">Hangup <a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a> [CS_NEW] [INCOMPATIBLE_DESTINATION]
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] switch_channel.c:1076
</pre>
</blockquote>
<pre wrap="">switch_channel_perform_hangup() Kill <a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a> [KILL]
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] switch_core_session.c:638
</pre>
</blockquote>
<pre wrap="">switch_core_session_signal_state_change() Kill
sofia//777888888@ a.b.c.d:5060 [BREAK]
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] sofia.c:71 sofia_event_callback() event
</pre>
</blockquote>
<pre wrap="">[nua_i_state] status [488][Not Acceptable Here] session:
<a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a>
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state()
</pre>
</blockquote>
<pre wrap="">Channel <a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a> entering state [terminated]
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] switch_core_state_machine.c:347
</pre>
</blockquote>
<pre wrap="">switch_core_session_run() (sofia//777888888@ a.b.c.d:5060) State HANGUP
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel
</pre>
</blockquote>
<pre wrap=""><a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a> hanging up, cause: INCOMPATIBLE_DESTINATION
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] switch_core_state_machine.c:45
</pre>
</blockquote>
<pre wrap="">switch_core_standard_on_hangup() Standard HANGUP
sofia//777888888@ a.b.c.d:5060, cause: INCOMPATIBLE_DESTINATION
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [DEBUG] switch_core_session.c:697
</pre>
</blockquote>
<pre wrap="">switch_core_session_thread() Session 1 (<a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a>)
Locked, Waiting on external entities
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [INFO] switch_core_session.c:703
</pre>
</blockquote>
<pre wrap="">switch_core_session_thread() Session 1 (<a class="moz-txt-link-abbreviated" href="mailto:sofia//777888888@a.b.c.d:5060">sofia//777888888@a.b.c.d:5060</a>) Ended
</pre>
<blockquote type="cite">
<pre wrap="">2007-09-20 10:40:24 [NOTICE] switch_core_session.c:705
</pre>
</blockquote>
<pre wrap="">switch_core_session_thread() Close Channel sofia//777888888@ a.b.c.d:5060
[CS_HANGUP]
</pre>
<blockquote type="cite">
<pre wrap="">
Which configuration file should i edit so i could pass the traffic from A
</pre>
</blockquote>
<pre wrap="">to B through fs?
</pre>
<blockquote type="cite">
<pre wrap="">Here's my config@fs :
default_context.xml
<context name="default">
<!--outgoing extension-->
<extension name="test1">
<condition field="destination_number"
</pre>
</blockquote>
<pre wrap="">expression="^(9991111[0-3]{3})$">
</pre>
<blockquote type="cite">
<pre wrap=""> <action application="set" data="call_timeout=30"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set"
</pre>
</blockquote>
<pre wrap="">data="hangup_after_bridge=true"/>
</pre>
<blockquote type="cite">
<pre wrap=""> <action application="bridge"
</pre>
</blockquote>
<pre wrap="">data=<a class="moz-txt-link-rfc2396E" href="mailto:sofia/gateway/test1/$1@w.x.y.z">"sofia/gateway/test1/$1@w.x.y.z"</a>/>
</pre>
<blockquote type="cite">
<pre wrap=""> </condition>
</extension>
<extension name="test2">
<condition field="destination_number"
</pre>
</blockquote>
<pre wrap="">expression="^(777888[0-9]{3})$">
</pre>
<blockquote type="cite">
<pre wrap=""> <action application="bridge"
</pre>
</blockquote>
<pre wrap="">data=<a class="moz-txt-link-rfc2396E" href="mailto:sofia/$${sip_profile}/$1@a.b.c.d">"sofia/$${sip_profile}/$1@a.b.c.d"</a>/>
</pre>
<blockquote type="cite">
<pre wrap=""> </condition>
</extension>
</context>
freeswitch.xml
<?xml version="1.0"?>
<document type="freeswitch/xml">
<!-- Preprocessor Variables
These are introduced when configuration strings must be consistent
</pre>
</blockquote>
<pre wrap="">across modules.
</pre>
<blockquote type="cite">
<pre wrap=""> -->
<!-- sip_profile
Must be a domain name if you are being a registry server; otherwise
can be any string.
used by: sofia.conf.xml enum.conf.xml default_context.xml
</pre>
</blockquote>
<pre wrap="">directory.xml
</pre>
<blockquote type="cite">
<pre wrap=""> -->
#set "sip_profile= 1.2.3.4"
<!-- xmpp_client_profile and xmpp_server_profile
xmpp_client_profile can be any string.
xmpp_server_profile is appended to "dingaling_" to form the
</pre>
</blockquote>
<pre wrap="">database name
</pre>
<blockquote type="cite">
<pre wrap=""> containing the "subscriptions" table.
used by: dingaling.conf.xml enum.conf.xml
-->
#set "global_codec_prefs=PCMU@20i,G729@20"
<!--#set "xmpp_client_profile=xmppc"-->
<!--#set "xmpp_server_profile=xmpps"-->
<!-- bind_server_ip
Can be an ip address, a dns name, or "auto".
This determines an ip address available on this host to bind.
If you are separating RTP and SIP traffic, you will want to have
use different addresses where this variable appears.
Used by: sofia.conf.xml dingaling.conf.xml
-->
<!--#set "bind_server_ip=auto"-->
<!-- external_rtp_ip
Used as the public IP address for SDP.
Can be an ip address or a string like "stun: stun.server.com"
If unspecified, the bind_server_ip value is used.
Used by: sofia.conf.xml dingaling.conf.xml
-->
<!--#set "external_rtp_ip=stun: stun.server.com"-->
<!-- server_name
A public ip address or DNS name that is used when advertising
</pre>
</blockquote>
<pre wrap="">conference
</pre>
<blockquote type="cite">
<pre wrap=""> presence or registering sip.
Used by: conference.conf.xml
-->
<!-- outbound_caller_id and outbound_caller_name
The caller ID telephone number we should use when calling out.
Used by: conference.conf.xml
-->
<!--#set "outbound_caller_name=FreeSWITCH"-->
<!--#set "outbound_caller_id=8777423583"-->
<section name="configuration" description="Various Configuration">
<!--#include "switch.conf.xml"-->
<!--#include "modules.conf.xml"-->
<!-- Order doesn't matter, but for clarity these are in same order as
</pre>
</blockquote>
<pre wrap="">modules.conf.xml.
</pre>
<blockquote type="cite">
<pre wrap=""> If they aren't loaded by modules.conf.xml, then they are ignored.
-->
<!-- Loggers -->
<!--#include "console.conf.xml"-->
<!--#include "syslog.conf.xml"-->
<!-- Multi-Faceted -->
<!--#include "enum.conf.xml"-->
<!-- XML Interfaces -->
<!--#include "xml_rpc.conf.xml"-->
<!--#include "xml_cdr.conf.xml"-->
<!--#include "xml_curl.conf.xml"-->
<!-- none for mod_xml_cdr -->
<!-- Event Handlers -->
<!--#include "cdr.conf.xml"-->
<!--#include "event_multicast.conf.xml"-->
<!--#include "event_socket.conf.xml"-->
<!--#include "xmpp_event.conf.xml"-->
<!--#include "zeroconf.conf.xml"-->
<!-- Directory Interfaces -->
<!-- none for mod_ldap; dialplan_directory.conf.xml has ldap
</pre>
</blockquote>
<pre wrap="">connection info -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- Endpoints -->
<!--#include "dingaling.conf.xml"-->
<!--#include " iax.conf.xml"-->
<!--#include "portaudio.conf.xml"-->
<!--#include "alsa.conf.xml"-->
<!--#include "sofia.conf.xml"-->
<!--#include " wanpipe.conf.xml"-->
<!--#include "woomera.conf.xml"-->
<!-- Applications -->
<!-- none for mod_bridgecall, mod_commands, mod_echo, mod_park,
</pre>
</blockquote>
<pre wrap="">mod_playback -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--#include "conference.conf.xml"-->
<!-- ivr.conf is used by mod_dptools -->
<!--#include "ivr.conf.xml"-->
<!-- Dialplan Interfaces -->
<!--#include "dialplan_directory.conf.xml"-->
<!-- mod_dialplan_xml is configured in the separate "dialplan"
</pre>
</blockquote>
<pre wrap="">section. -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- Codec Interfaces -->
<!-- no configuration needed -->
<!-- File Format Interfaces -->
<!-- no configuration needed -->
<!-- Timers -->
<!-- no configuration needed -->
<!-- Languages -->
<!--#include " spidermonkey.conf.xml"-->
<!-- none for mod_perl -->
<!-- ASR /TTS -->
<!-- none for mod_cepstral -->
<!--#include "rss.conf.xml"-->
<!--#include "mod_openmrcp.conf.xml"-->
<!-- Say -->
<!-- none for mod_say_en -->
<!--#include "mod_cdr.conf.xml"-->
<!--#include "mod_local_stream.conf.xml"-->
</section>
<section name="dialplan" description="Regex/XML Dialplan">
<!--#include "default_context.xml"-->
</section>
<!-- mod_dingaling is reliant on the vcard data in the "directory"
</pre>
</blockquote>
<pre wrap="">section. -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- mod_sofia is reliant on the user data for authorization -->
<section name="directory" description="User Directory">
<!--#include "directory.xml"-->
</section>
<!-- phrases section (under development still) -->
<section name="phrases" description="Speech Phrase Management">
<macros>
<language name="en" sound_path="/snds" tts_engine="cepstral"
</pre>
</blockquote>
<pre wrap="">tts_voice="david">
</pre>
<blockquote type="cite">
<pre wrap=""> <!--#include "lang_en.xml"-->
</language>
<language name="fr"
</pre>
</blockquote>
<pre wrap="">sound_path="/var/sounds/lang/fr/jean" tts_engine="cepstral"
tts_voice="jean-pierre">
</pre>
<blockquote type="cite">
<pre wrap=""> <!--#include "lang_fr.xml"-->
</language>
</macros>
</section>
</document>
Sofia.conf.xml
<configuration name=" sofia.conf" description="sofia Endpoint">
<profiles>
<profile name="test1">
<!--aliases are other names that will work as a valid profile name
</pre>
</blockquote>
<pre wrap="">for this profile-->
</pre>
<blockquote type="cite">
<pre wrap=""> <aliases>
<alias name="test1"/>
</aliases>
<!-- Outbound Registrations -->
<gateways>
<gateway name="test1">
<!--/// account username *required* ///-->
<param name="username" value="myusername B"/>
<!--/// auth realm: *optional* same as gateway name, if blank
</pre>
</blockquote>
<pre wrap="">///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="realm" value="1.2.3.4"/>
<!--/// domain to use in from: *optional* same as realm, if
</pre>
</blockquote>
<pre wrap="">blank ///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="from-domain" value="asterlink.com"/>-->
<!--/// account password *required* ///-->
<param name="password" value="xxxx"/>
<!--/// replace the INVITE from user with the channel's
</pre>
</blockquote>
<pre wrap="">caller-id ///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="caller-id-in-from" value="false"/>-->
<!--/// extension for inbound calls: *optional* same as
</pre>
</blockquote>
<pre wrap="">username, if blank ///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="extension" value="myusername B"/>
<!--/// proxy host: *optional* same as realm, if blank ///-->
<param name="proxy" value="1.2.3.4"/>
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<param name="expire-seconds" value="60"/>
<!--/// do not register ///-->
<param name="register" value="true"/>
<!--How many seconds before a retry when a failure or timeout
</pre>
</blockquote>
<pre wrap="">occurs -->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="retry_seconds" value="30"/>
<!--Use the callerid of an inbound call in the from field on
</pre>
</blockquote>
<pre wrap="">outbound calls via this gateway -->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="disable-transcoding" value="true"/>
<param name="caller-id-in-from" value="false"/>
</gateway>
</gateways>
<domains>
<!-- indicator to parse the directory for domains with
</pre>
</blockquote>
<pre wrap="">parse="true" to get gateways-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<domain name="$${domain}" parse="true"/>-->
</domains>
<settings>
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs"
</pre>
</blockquote>
<pre wrap="">value="$${global_codec_prefs}"/>
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="codec-ms" value="20"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="$${bind_server_ip}"/>
<param name="sip-ip" value="$${bind_server_ip}"/>
<!--set to 'greedy' if you want your codec list to take precedence
</pre>
</blockquote>
<pre wrap="">-->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="inbound-codec-negotiation"
</pre>
</blockquote>
<pre wrap="">value="generous"/>
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--If you don't want to pass through timestampes from 1 RTP call
</pre>
</blockquote>
<pre wrap="">to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="rtp-rewrite-timestampes" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it
</pre>
</blockquote>
<pre wrap="">instead of SQLite-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-no-media" value="true"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if
</pre>
</blockquote>
<pre wrap="">the codec is ok-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="inbound-late-negotiation" value="true"/>-->
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other
</pre>
</blockquote>
<pre wrap="">2 lines for call authentication -->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="accept-blind-reg" value="true"/>
<!--TTL for nonce in sip auth-->
<param name="nonce-ttl" value="60"/>
<!--Uncomment if you want to force the outbound leg of a bridge to
</pre>
</blockquote>
<pre wrap="">only offer the codec
</pre>
<blockquote type="cite">
<pre wrap=""> that the originator is using-->
<!--<param name="disable-transcoding" value="true"/>-->
<!--<param name="auth-calls" value="true"/>-->
<!-- on authed calls, authenticate *all* the packets not just
</pre>
</blockquote>
<pre wrap="">invite -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="auth-all-packets" value="true"/>-->
<!-- <param name="ext-rtp-ip"
</pre>
</blockquote>
<pre wrap="">value="$${external_rtp_ip}"/>-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- <param name="ext-sip-ip" value="100.101.102.103"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
</settings>
</profile>
<profiles>
</profiles>
<profile name="test2">
<!--aliases are other names that will work as a valid profile name
</pre>
</blockquote>
<pre wrap="">for this profile-->
</pre>
<blockquote type="cite">
<pre wrap=""> <aliases>
<alias name="test2"/>
</aliases>
<gateways>
<gateway name="test2">
<!--/// account username *required*///-->
<param name="username" value="username A"/>
<!--/// auth realm: *optional* same as gateway name, if blank
</pre>
</blockquote>
<pre wrap="">///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="realm" value=" a.b.c.d"/>
<!--/// domain to use in from: *optional* same as realm, if
</pre>
</blockquote>
<pre wrap="">blank ///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="from-domain" value=" asterlink.com"/>-->
<!--/// account password *required* ///-->
<param name="password" value="password"/>
<!--/// replace the INVITE from user with the channel's
</pre>
</blockquote>
<pre wrap="">caller-id ///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="caller-id-in-from" value="false"/>
<!--/// extension for inbound calls: *optional* same as
</pre>
</blockquote>
<pre wrap="">username, if blank ///-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<param name="proxy" value="a.b.c.d"/>
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<param name="expire-seconds" value="60"/>
<!--/// do not register ///-->
<param name="register" value="true"/>
<!--How many seconds before a retry when a failure or timeout
</pre>
</blockquote>
<pre wrap="">occurs -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="retry_seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on
</pre>
</blockquote>
<pre wrap="">outbound calls via this gateway -->
</pre>
<blockquote type="cite">
<pre wrap=""> -<param name="caller-id-in-from" value="false"/>
<param name="disable-transcoding" value="true"/>
</gateway>
</gateways>
<settings>
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5061"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs"
</pre>
</blockquote>
<pre wrap="">value="$${global_codec_prefs}"/>
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="codec-ms" value="20"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="$${bind_server_ip}"/>
<param name="sip-ip" value="$${bind_server_ip}"/>
<!--set to 'greedy' if you want your codec list to take precedence
</pre>
</blockquote>
<pre wrap="">-->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="inbound-codec-negotiation"
</pre>
</blockquote>
<pre wrap="">value="generous"/>
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--If you don't want to pass through timestampes from 1 RTP call
</pre>
</blockquote>
<pre wrap="">to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="rtp-rewrite-timestampes" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it
</pre>
</blockquote>
<pre wrap="">instead of SQLite-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-no-media" value="true"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if
</pre>
</blockquote>
<pre wrap="">the codec is ok-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="inbound-late-negotiation" value="true"/>-->
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other
</pre>
</blockquote>
<pre wrap="">2 lines for call authentication -->
</pre>
<blockquote type="cite">
<pre wrap=""> <param name="accept-blind-reg" value="true"/>
<!--TTL for nonce in sip auth-->
<param name="nonce-ttl" value="60"/>
<!--Uncomment if you want to force the outbound leg of a bridge to
</pre>
</blockquote>
<pre wrap="">only offer the codec
</pre>
<blockquote type="cite">
<pre wrap=""> that the originator is using-->
<!--<param name="disable-transcoding" value="true"/>-->
<!--<param name="auth-calls" value="true"/>-->
<!-- on authed calls, authenticate *all* the packets not just
</pre>
</blockquote>
<pre wrap="">invite -->
</pre>
<blockquote type="cite">
<pre wrap=""> <!--<param name="auth-all-packets" value="true"/>-->
<!-- <param name="ext-rtp-ip"
</pre>
</blockquote>
<pre wrap="">value="$${external_rtp_ip}"/>-->
</pre>
<blockquote type="cite">
<pre wrap=""> <!-- <param name="ext-sip-ip" value="100.101.102.103"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!-- <param name="ext-sip-ip" value=" 100.101.102.103"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value=<a class="moz-txt-link-rfc2396E" href="sip:10.0.1.251:5555">"sip:10.0.1.251:5555"</a>/>-->
</settings>
</profile>
</profiles>
</configuration>
appreciate it if anybody could give me clue
Thx,
~pieter~
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