June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to pass a call from one FS to another FS ??
by Brad Tuan 02 Jun '09

02 Jun '09

Like this?? *> <extension name="remoteFreeswitch"> *>* <condition field="destination_number" expression="^014(\d+)$"> *>* *<action application="bridge" data="sofia/${use_profile}/$1%192.168.141.187"/> >* </condition> *>* </extension> * I have tried,but Fs still return: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001(a)192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001(a)192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED]

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How to specify the path to fax file on Windows?
by Paul Li 02 Jun '09

02 Jun '09

All the examples related to faxing use the Unix/Linux path, such as originate sofia/external/100(a)10.10.10.10 &txfax(/path_to_fax_file) <extension name="fax_transmit"> <condition field="destination_number" expression="^9979$"> <action application="txfax" data="/tmp/txfax.tiff"/> <action application="hangup"/> </condition> </extension> I have tried "C:/tmp/fax/txfax.tiff" or "C:\MyJob\fax\txfax.tiff" or "C:\\MyJob\\fax\\txfax.tiff" without any luck. I got an error like [ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file, or the app crashed. Any help is greatly appreciated!

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How to pass a call from one FS to another FS ??
by Brad Tuan 02 Jun '09

02 Jun '09

I have tried <extension name="remoteFreeswitch"> <condition field="destination_number" expression="^014(\d+)$"> <action application="bridge" data="sofia/external/$1 at 192.168.141.187<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> :5080"/> </condition> </extension> But the console moniter return : [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187:5080@external [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED I am sure that 97710001 is a online user on "192.168.141.187", What's wrong??

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How to pass a call from one FS to another FS ??
by Brad Tuan 02 Jun '09

02 Jun '09

>>* This question can be separated into two part: *>>* 1.Pass a call to another FS * >uuid_deflect or uuid_transfer, depending on whether the call has been answered >by the first FS instance or not. See the wiki. >>* *>>* 2.Receive a call from another FS * >Provide a dial plan entry in the second FS that handles the call appropriately >after the deflect or transfer. >You haven't explained what you're trying to do - a general question warrants a >general answer. I am assuming the call is arriving at one FS system, and >(before or after answering it), you want to move it to another FS system. >That's the question I've answered above. The wiki documents the syntax. SorrySorry,let me explain my question. When User1( User of FS1 ) call User2( User of FS2 ) , FS1 will pass the call to FS2 before answering, and then User1 can talk with User2. -Brad

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Polycom Phone issue
by Henry Huang 02 Jun '09

02 Jun '09

I have setup 2 Freeswitch test server, 2 IP430 polycom phones with 2 lines each registered to different server. When I dial extension from the first line (first FS server), if the called party hit reject on the phone. The called won't be hanged up will keep ringing until timed out. When dialed from second line (2nd FS server), the other phone would ring on the screen as of the first line is ringing and besides the caller-id and caller-id-name, it will also show the sip: address of the caller. Now it has similar sympton to the above senario, but more over, even if leg B picks up the call then hang up, Leg A (caller's end) will not hang up for as long as you don't hang it up. Do any of you people with polycom phone have problem like this ? or know what could be the cause? my FS1 server is version 12242M , FS2 server version is 13523M *Here is my dialplan example of the extension that I called:* <extension name="eric"> <condition field="destination_number" expression="^(2025)$"/> <condition field="caller_id_number" expression="^(\d{2})(\d{2})$"> <action application="set" data="caller_id_number=some_number"/> <action application="set" data="call_timeout=35"/> <action application="bridge" data="user/2025@ ${domain_name},openzap/1/a/some_number"/> <anti-action application="set" data="caller_id_number=some_number"/> <anti-action application="set" data="call_timeout=35"/> <anti-action application="bridge" data="user/2025@ ${domain_name},openzap/1/a/some_number"/> </condition> </extension> * And here is the setting for register 2 server on polycom phone:* reg.1.displayName="2025" reg.1.address="2025" reg.1.label="2025" reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="" reg.1.auth.userId="2025" reg.1.auth.password="somepassword" reg.1.server.1.address="10.48.5.83" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.expires.overlap="" reg.1.server.1.register="1" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.outboundProxy.address="" reg.1.outboundProxy.port="" reg.1.outboundProxy.transport="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.proxyRequire="" reg.1.ringType="12" reg.1.lineKeys="1" reg.1.callsPerLineKey="" reg.2.displayName="2025" reg.2.address="2025" reg.2.label="2025" reg.2.type="private" reg.2.lcs="" reg.2.thirdPartyName="" reg.2.auth.userId="2025" reg.2.auth.password="somepassword" reg.2.server.1.address="10.55.7.36" reg.2.server.1.port="5060" reg.2.server.1.transport="DNSnaptr" reg.2.server.2.transport="DNSnaptr" reg.2.server.1.expires="" reg.2.server.1.expires.overlap="" reg.2.server.1.register="" reg.2.server.1.retryTimeOut="" reg.2.server.1.retryMaxCount="" reg.2.server.1.expires.lineSeize="" reg.2.outboundProxy.address="" reg.2.outboundProxy.port="" reg.2.outboundProxy.transport="" reg.2.acd-login-logout="0" reg.2.acd-agent-available="0" -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant

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How to pass a call from one FS to another FS ??
by Brad Tuan 02 Jun '09

02 Jun '09

This question can be separated into two part: 1.Pass a call to another FS 2.Receive a call from another FS Somebody can tell me how to do these?? Please............. -Brad

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Conference users hear MOH until leader enters?
by Jason Cromes 02 Jun '09

02 Jun '09

First off, I apologize if this has been sent multiple times, the mailing list won't cooperate with me... Hopefully that is resolved now. I'm attempting to replicate the behavior of an Asterisk conferencing system and I need a feature that, I'm surprised to say, doesn't seem to be supported (easily) in FreeSwitch. I've setup my dialplan so that when a specific extension is hit, it calls out to some javascript which acts like an IVR to handle the prompts and conferencing setup. (Similar to this: http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own improvements.) Anyway, the conferences are stored permanently in a database table, but I need them disabled until the conference owner dials in and enables his conference. If other users have entered the conference prior to the owner, they should hear music-on-hold until the leader enters. (The purpose of this is to prevent abuse of conferences that are, basically, always available.) This is easy in Asterisk because you can pop someone into MeetMe with different flags. So, in my IVR, I prompt for the "conference number" (known to all) and then the "password" (known only to the owner/leader). If the proper password is entered, that user is sent to conference XYZ with the leader flag set. If no password is entered, the user goes to conference XYZ, without the leader flag. If anyone enters before the leader, they're told by MeetMe that "the conference will begin when the leader arrives" and MeetMe provides MOH until that time. Help! This is an absolute deal-breaker for my install... How can I do this in FreeSwitch? Thanks...

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Re: [Freeswitch-users] Passthru mode
by FERNANDO VILLARROEL 01 Jun '09

01 Jun '09

Hello i was try with: <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/gateway/ubb/$1$2$3"/> This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins <msc(a)freeswitch.org> wrote: > From: Michael Collins <msc(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users(a)lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL <fvillarroel(a)yahoo.com> > wrote: > > > > Hello the dial plan: > > > > <action application="bridge" > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > <action application="set" > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > <param name="inbound-proxy-media" > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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Re: [Freeswitch-users] Passthru mode
by FERNANDO VILLARROEL 01 Jun '09

01 Jun '09

Hello i was try with: <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/gateway/ubb/$1$2$3"/> This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins <msc(a)freeswitch.org> wrote: > From: Michael Collins <msc(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users(a)lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL <fvillarroel(a)yahoo.com> > wrote: > > > > Hello the dial plan: > > > > <action application="bridge" > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > <action application="set" > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > <param name="inbound-proxy-media" > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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Re: [Freeswitch-users] Passthru mode
by FERNANDO VILLARROEL 01 Jun '09

01 Jun '09

Hello the dial plan: <action application="bridge" data="sofia/gateway/ubb/$1$2$3"/> This i setup from Wikipbx. --- On Mon, 6/1/09, Michael Collins <msc(a)freeswitch.org> wrote: > From: Michael Collins <msc(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users(a)lists.freeswitch.org > Date: Monday, June 1, 2009, 2:15 PM > What does your dialplan look like? Just > curious where/how you set proxy-media mode. > -MC > > On Mon, Jun 1, 2009 at 9:54 AM, > FERNANDO VILLARROEL <fvillarroel(a)yahoo.com> > wrote: > > > > Dear all. > > > > I have problem with g729 passthru mode. > > > > I received traffic from a Asterisk on my FS and forward to > other Asterisk, when i use codec ulaw this works very well. > > > > But when i try use G729 i received the following messages > and SIP Trace: > > > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel > sofia/admin/42452904(a)190.208.xx.yy > [f65514e0-4ec7-11de-9b78-150e2985561f] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 > sofia_handle_sip_i_state() Channel > sofia/admin/42452904(a)190.208.xx.yy entering state > [received][100] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 > sofia_handle_sip_i_state() Remote SDP: > > v=0 > > o=root 25643 25643 IN IP4 190.208.xx.yy > > s=session > > c=IN IP4 190.208.xx.yy > > t=0 0 > > m=audio 10236 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [G729:18:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 > sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 > sofia_handle_sip_i_state() Hangup > sofia/admin/42452904(a)190.208.xx.yy [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904(a)190.208.xx.yy [KILL] > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904(a)190.208.xx.yy [BREAK] > > send 886 bytes to udp/[190.47.91.83]:60245 at > 16:19:20.596633: > > > ------------------------------------------------------------------------ > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > > From: "102" > <sip:102@200.111.XXX.XX>;tag=bmke36jc1v > > To: "102" > <sip:102@200.111.XXX.XX>;tag=rZp0XXrK9NHFD > > Call-ID: > 3c26700b249f-sryanqz0td8u@snom360-00041323143F > > CSeq: 27056 REGISTER > > Contact: > <sip:102@192.168.1.124:2051;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:72e96588-ebe1-476d-8024-75656b4e007d>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > > > Date: Mon, 01 Jun 2009 16:19:20 GMT > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904(a)190.208.xx.yy) Running State Change > CS_HANGUP > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904(a)190.208.xx.yy) State HANGUP > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 > sofia_on_hangup() Channel sofia/admin/42452904(a)190.208.xx.yy > hanging up, cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 > sofia_on_hangup() Responding to INVITE with: 488 > > send 634 bytes to udp/[190.208.xx.yy]:5060 at > 16:19:20.603208: > > > ------------------------------------------------------------------------ > > SIP/2.0 488 Not Acceptable Here > > Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > > From: "452904" > <sip:42452904@190.208.xx.yy>;tag=as4e2616ae > > To: > <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r > > Call-ID: > 117330d21f3828470f39a95f538be036(a)190.208.xx.yy > > CSeq: 102 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, refer > > Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() > sofia/admin/42452904(a)190.208.xx.yy Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904(a)190.208.xx.yy) State HANGUP going to > sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() > (sofia/admin/42452904(a)190.208.xx.yy) State Change CS_HANGUP > -> CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904(a)190.208.xx.yy [BREAK] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904(a)190.208.xx.yy) Running State Change > CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904(a)190.208.xx.yy) State REPORTING > > recv 408 bytes from udp/[190.208.xx.yy]:5060 at > 16:19:20.620338: > > > ------------------------------------------------------------------------ > > ACK sip:56968482060@200.111.XXX.XX SIP/2.0 > > Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > > From: "452904" > <sip:42452904@190.208.xx.yy>;tag=as4e2616ae > > To: > <sip:56968482060@200.111.XXX.XX>;tag=S8FSZr9p6y71r > > Contact: <sip:42452904@190.208.xx.yy> > > Call-ID: > 117330d21f3828470f39a95f538be036(a)190.208.xx.yy > > CSeq: 102 ACK > > User-Agent: Asterisk PBX > > Max-Forwards: 70 > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() > sofia/admin/42452904(a)190.208.xx.yy Standard REPORTING, > cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904(a)190.208.xx.yy) State REPORTING going > to sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() > (sofia/admin/42452904(a)190.208.xx.yy) State Change > CS_REPORTING -> CS_DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 > (sofia/admin/42452904(a)190.208.xx.yy) Locked, Waiting on > external entities > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 > (sofia/admin/42452904(a)190.208.xx.yy) Ended > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904(a)190.208.xx.yy [CS_DESTROY] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904(a)190.208.xx.yy) State DESTROY > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 > sofia_on_destroy() sofia/admin/42452904(a)190.208.xx.yy SOFIA > DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() > sofia/admin/42452904(a)190.208.xx.yy Standard DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904(a)190.208.xx.yy) State DESTROY going to > sleep > > > > My vars.xml : > > > > <X-PRE-PROCESS cmd="set" > data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > <X-PRE-PROCESS cmd="set" > data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > <X-PRE-PROCESS cmd="set" > data="xmpp_client_profile=xmppc"/> > > <X-PRE-PROCESS cmd="set" > data="xmpp_server_profile=xmpps"/> > > > > > > I hope your comments for know where is the config problem > > > > Fernando. > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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