sofia//777888888@*a.b.c.d*:5060 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 <brian.west@mac.com> wrote:
Not Acceptable Here would indicate aa codec isssue. As would INCOMPATIBLE_DESTINATION as the hangup cause. Can you verify the codec configs on each end? /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 between gateways but am still stuck with the proper configuration. The simple diagram is like this :
*subscriber A------> SIP Server A ------>Freeswitch----->SIP Server B-----> subscriber B* voip prefix ip a.b.c.d ip 1.2.3.4 ip w.x.y.z voip prefix 777888x 999111x
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() Hangup sofia//777888888@*a.b.c.d*:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2007-09-20 10:40:24 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia//777888888@*a.b.c.d:*5060 [KILL] 2007-09-20 10:40:24 [DEBUG] switch_core_session.c:638 switch_core_session_signal_state_change() Kill sofia//777888888@*a.b.c.d*:5060 [BREAK] 2007-09-20 10:40:24 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_state] status [488][Not Acceptable Here] session: sofia//777888888@ *a.b.c.d*:5060 2007-09-20 10:40:24 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Channel sofia//777888888@*a.b.c.d:*5060 entering state [terminated] 2007-09-20 10:40:24 [DEBUG] switch_core_state_machine.c:347 switch_core_session_run() (sofia//777888888@*a.b.c.d*:5060) State HANGUP 2007-09-20 10:40:24 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel sofia//777888888@*a.b.c.d*:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2007-09-20 10:40:24 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia//777888888@*a.b.c.d *:5060, cause: INCOMPATIBLE_DESTINATION 2007-09-20 10:40:24 [DEBUG] switch_core_session.c:697 switch_core_session_thread() Session 1 (sofia//777888888@*a.b.c.d*:5060) Locked, Waiting on external entities 2007-09-20 10:40:24 [INFO] switch_core_session.c:703 switch_core_session_thread() Session 1 (sofia//777888888@*a.b.c.d*:5060) Ended 2007-09-20 10:40:24 [NOTICE] switch_core_session.c:705 switch_core_session_thread() Close Channel sofia//777888888@*a.b.c.d*:5060 [CS_HANGUP]
Which configuration file should i edit so i could pass the traffic from A to B through fs?
Here's my config@fs :
*default_context.xml
*<context name="default">
<!--outgoing extension--> <extension name="test1"> <condition field="destination_number" expression="^(9991111[0-3]{3})$"> <action application="set" data="call_timeout=30"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/test1/$1@* w.x.y.z*"/> </condition> </extension>
<extension name="test2"> <condition field="destination_number" expression="^(777888[0-9]{3})$"> <action application="bridge" data="sofia/$${sip_profile}/$1@* a.b.c.d*"/> </condition> </extension>
</context>
*freeswitch.xml*
<?xml version="1.0"?> <document type="freeswitch/xml">
<!-- Preprocessor Variables These are introduced when configuration strings must be consistent across modules. --> <!-- 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 directory.xml --> #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 database name 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 conference 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 modules.conf.xml. 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 connection info -->
<!-- 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, mod_playback --> <!--#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" section. -->
<!-- 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" section. --> <!-- 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" tts_voice="david"> <!--#include "lang_en.xml"--> </language> <language name="fr" sound_path="/var/sounds/lang/fr/jean" tts_engine="cepstral" tts_voice="jean-pierre"> <!--#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 for this profile--> <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 ///--> <param name="realm" value="*1.2.3.4*"/> <!--/// domain to use in from: *optional* same as realm, if blank ///--> <!--<param name="from-domain" value="asterlink.com"/>--> <!--/// account password *required* ///--> <param name="password" value="xxxx"/> <!--/// replace the INVITE from user with the channel's caller-id ///--> <!--<param name="caller-id-in-from" value="false"/>--> <!--/// extension for inbound calls: *optional* same as username, if blank ///--> <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 occurs --> <param name="retry_seconds" value="30"/> <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> <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 parse="true" to get gateways--> <!--<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" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/> <!--set to 'greedy' if you want your codec list to take precedence --> <param name="inbound-codec-negotiation" value="generous"/> <!-- if you want to send any special bind params of your own --> <!--<param name="bind-params" value="transport=udp"/>-->
<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)--> <!--<param name="rtp-rewrite-timestampes" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite--> <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--Uncomment to set all inbound calls to no media mode--> <!--<param name="inbound-no-media" value="true"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> <!--<param name="inbound-late-negotiation" value="true"/>-->
<!-- this lets anything register --> <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!--TTL for nonce in sip auth--> <param name="nonce-ttl" value="60"/> <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec that the originator is using--> <!--<param name="disable-transcoding" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!-- on authed calls, authenticate *all* the packets not just invite --> <!--<param name="auth-all-packets" value="true"/>-->
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
<!-- <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 for this profile--> <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 ///--> <param name="realm" value="*a.b.c.d*"/> <!--/// domain to use in from: *optional* same as realm, if blank ///--> <!--<param name="from-domain" value="asterlink.com"/>--> <!--/// account password *required* ///--> <param name="password" value="*password*"/> <!--/// replace the INVITE from user with the channel's caller-id ///--> <param name="caller-id-in-from" value="false"/> <!--/// extension for inbound calls: *optional* same as username, if blank ///--> <!--<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 occurs --> <!--<param name="retry_seconds" value="30"/>--> <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> -<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" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/> <!--set to 'greedy' if you want your codec list to take precedence --> <param name="inbound-codec-negotiation" value="generous"/> <!-- if you want to send any special bind params of your own --> <!--<param name="bind-params" value="transport=udp"/>-->
<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)--> <!--<param name="rtp-rewrite-timestampes" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite--> <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--Uncomment to set all inbound calls to no media mode--> <!--<param name="inbound-no-media" value="true"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> <!--<param name="inbound-late-negotiation" value="true"/>-->
<!-- this lets anything register --> <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!--TTL for nonce in sip auth--> <param name="nonce-ttl" value="60"/> <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec that the originator is using--> <!--<param name="disable-transcoding" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!-- on authed calls, authenticate *all* the packets not just invite --> <!--<param name="auth-all-packets" value="true"/>-->
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
<!-- <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="sip:10.0.1.251:5555"/>--> </settings> </profile> </profiles> </configuration>
appreciate it if anybody could give me clue
Thx,
~pieter~
*
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