January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Overlap dialing problem
by Štefan Čudai 10 Jan '12

10 Jan '12

Hi, does FreeSwitch support for overlap dialing or did somebody try to use it? I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured my freetdm.conf.xml with this two lines: <param name="overlap" value="yes"/> <param name="min_digits" value="0"/> But when I'm dialing number (which has 14 digits) I hear disconnect tone after dialing first 5 digits. I think that reason is because FreeTDM module is waiting number with minimum digits equals to "0" (see my min_digits). I have also try set min_digits to 20 and then dial same number with 14 digits but result was same. So, I looked to log file and found that FreeTDM module got in thist case STATUS CONFIRM message but he was still waiting for next digits and his time expired. Can you help me please? Thanks a lot. Stefan

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Calling multiple phones at once
by Ryan How 10 Jan '12

10 Jan '12

Hi, How do I set it up in freeswitch to call mulitple phones at once (internal and via a gateway) that works with call_timeout. I can't seem to find the appropriate wiki page. Call groups kinda work, but then the phones need to register. My setup with bridge the timeout doesn't work <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/1000(a)$${domain},sofia/gateway/sip.pennytel.com/0447123456"/> <action application="answer"/> <action application="bridge" data="loopback/app=voicemail:default $${domain} 1000"/> <action application="hangup"/> After 30 seconds all the phones start ringing again. And when the external number hangs up it immediately starts ringing again. With the call groups it seems to work correctly. Also, using the user/1000@domain notation, multiple registrations doesn't work, only a single device rings. Thanks!

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CISCO 2811 Freeswitch IVR
by Oliver Schenk 10 Jan '12

10 Jan '12

I've been battling while creating an IVR using FreeSWITCH mod_managed and connecting through a CISCO 2811. Most things now work quite well, but I am having a few issues with the way the system answers calls (or doesn't answer calls...). I have FreeSWITCH running as a windows service on Windows Server 2008, which is connected via LAN to a CISCO 2811 with a 4 port FXO card, which is then connected to a POTS phone line. Take the following scenario: 1. Managed .NET application creates a call string and uses ESL to talk to freeswitch and originate a call: string callstring = "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567(a)192.168.x.x '&managed(ivrAppName)'"; eslConnection.API("originate", callstring); where 192.168.x.x is the CISCO IP. 2. The CISCO sees that the phone number (1091234567) starts with a 1 so it uses FXO port 1 and strips the 1 and uses the remaining phone number (091234567) to make the call. 3. My phone rings, I pick up and I can hear my IVR playing. These are my current problems: - IVR starts playing before I even pick up the phone. This means that if the system calls a mobile phone and the person doesn't pick up, the IVR will start playing and eventually the mobile phone will divert to voice mail. Obviously I then get a missed call and an sms saying I have a new voice mail, which is annoying. Instead I would like it to KNOW that no one has picked up, but I don't know how to do this. Somehow the CISCO needs to be able to tell FreeSWITCH that the call has not yet been answered. For some reason however as soon as the CISCO starts calling FreeSWITCH thinks the call is already connected. It doesn't know that the CISCO is actually still ringing. Maybe I'm doing originate the wrong way or something ... - The phone only rings for about 10 seconds before hanging up. I've tried "call_timeout", "bridge_answer_timeout". I've also tried setting CISCO "ring number". Nothing works, my phone still only rings for about 10 seconds. I don't know if this is a FreeSWITCH issue or a CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just starts playing even if no one answers the phone. CISCO Config for relevant FXO port: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 supplementary-service h450.12 no supplementary-service sip moved-temporarily no supplementary-service sip refer fax protocol cisco sip registrar server expires max 3600 min 3600 no update-callerid no call service stop voice-port 0/3/2 output attenuation -3 no comfort-noise cptone AU impedance complex1 caller-id enable ! dial-peer voice 100 pots preference 1 destination-pattern 1T port 0/3/2 ! Many Thanks, Oliver

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(no subject)
by bwibowo＠gmail.com 10 Jan '12

10 Jan '12

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NAT Confusion
by Bob Smith 10 Jan '12

10 Jan '12

Hi, Newbie here (again !) .... Hope someone here can help. I'm having a few NAT related issues and the sparse, sometimes out-of-date wikis on the freeswitch site are not helping.... ;-( My layout is as follows ... Freewitch Server .....**INTERNET**........[FIREWALL].....[BRIA 3 SOFTPHONE] (Public IP, No NAT) .......................................NAT..............Internal IP Firewall traversal method in Bria is set to "none" as I've no ICE/STUN/TURN server. Freeswitch is started with the -nonat -nonatmap options because it is on a public IP with no NAT. (1) Outbound Calls I can dial out. I can hear the remote end. The remote end cannot hear me. (2) Inbound Calls These do not work at all. Console output (excluding debug level messages, and with IP and Phone Numbers changed to protect the innocent) 2012-01-09 19:22:14.076501 [NOTICE] switch_channel.c:924 New Channel sofia/internal/02079460949(a)10.0.0.134 [3c4ec25c-3af7-11e1-96ae-155d65675ea5] 2012-01-09 19:22:14.076501 [INFO] mod_dialplan_xml.c:481 Processing 02079460949 <02079460949>->01174960949 in context public 2012-01-09 19:22:14.096468 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/02079460949(a)10.0.0.134 to XML[1000@default] 2012-01-09 19:22:14.096468 [INFO] mod_dialplan_xml.c:481 Processing 02079460949 <02079460949>->1000 in context default 2012-01-09 19:22:14.096468 [NOTICE] switch_channel.c:924 New Channel sofia/internal/sip:1000@10.14.2.2:48101 [3c4f8cb4-3af7-11e1-96b5-155d65675ea5] 2012-01-09 19:22:14.096468 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-09 19:22:14.096468 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-09 19:22:14.156503 [NOTICE] sofia.c:6246 Hangup sofia/internal/sip:1000@10.14.2.2:48101 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] 2012-01-09 19:22:14.156503 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED] 2012-01-09 19:22:14.156503 [INFO] mod_dptools.c:2900 Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED 2012-01-09 19:22:14.156503 [NOTICE] switch_core_session.c:1398 Session 27 (sofia/internal/sip:1000@10.14.2.2:48101) Ended 2012-01-09 19:22:14.156503 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1000@10.14.2.2:48101 [CS_DESTROY] 2012-01-09 19:22:14.156503 [NOTICE] mod_dptools.c:3019 Hangup sofia/internal/02079460949(a)10.0.0.134 [CS_EXECUTE] [SERVICE_NOT_IMPLEMENTED] 2012-01-09 19:22:14.176780 [NOTICE] switch_core_session.c:1398 Session 26 (sofia/internal/02079460949(a)10.0.0.134) Ended 2012-01-09 19:22:14.176780 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/02079460949(a)10.0.0.134 [CS_DESTROY] The above would suggest I need to boot Freeswitch with NAT turned on ? But I'm confused as to why ? Because Freeswitch has no need to discover its public IP address. Thanks Bob

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FreeSWITCH Conference Call This Week: Sangoma
by Michael Collins 09 Jan '12

09 Jan '12

Greetings all! This week's conference call will feature Moises Silva from Sangoma. Moises (IRC: moy) has lots of experience with TDM communications and is the perfect person to present information on configuring PRI cards and FreeTDM. He also has some interesting information about new GSM stuff from Sangoma. We look forward to his presentation this coming Wednesday, Jan 11. The agenda page is in the usual place<http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_11>on the wiki. Talk to you all on Wednesday! -Michael

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Need Less CPU Intensive Way to Detect Voicemail and Answering Machines
by David Stein 09 Jan '12

09 Jan '12

Hello. I need to detect answering machine beeps on 1,000+ channels simultaneously. Unfortunately, this is a real requirement, not a pseudo-requirement that can be engineered away. This is for a mass emergency notification system for government, K-12, and higher education. In the past, I have done this with proprietary hardware (Dialogic and Aculab), which handles this sort of thing without overloading the host CPU, as the hardware's DSPs handle the detections. I want to see if I can do the same thing with FreeSWITCH. Alas, I find the following warning in the wiki page for mod_avmd: AVMD (and VMD) are both very CPU intensive. You need to be aware of this fact when using it. It will drastically reduce your call capacity if you do not manage it correctly. On the other hand it is a very useful tool, and if managed properly will be a great aid for calls needing to do Voice Mail Detection. Eric states, "You can expect about ~50 simultaneous instances on an Intel i7 920 CPU." So, it sounds like VMD and AVMD won't work for what I need to do. I also don't think that the tone_detect application will work, as this requires specific frequencies (as opposed to ranges of frequencies), and answering machines and voicemail come with beeps at all sorts of frequencies. Does anyone know anything, either free or commercial, that I can use with FreeSWITCH to do this many simultaneous detections? I know of commercial software-only platforms (e.g., Aculab's Prosody S) that claim to be able to this, so it seems like it should be possible. Thanks in advance, David Stein

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Max calls Limit?
by Anita Hall 09 Jan '12

09 Jan '12

Is there any config variable somewhere which is limiting the max call limit on FS to 1000? Set-up: Taking FS for max call load on my machine. User freeswitch is calling internal gateway and playing demo-ivr User freeswitch defined in vars.xml <X-PRE-PROCESS cmd="set" data="default_provider=10.60.20.145"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=freeswitch"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=freeswitch"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=10.60.20.145"/>  <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> Internal Gateway defined in directory/default/freeswitch.xml <include> <user id="freeswitch"> <params> <param name="password" value="freeswitch"/> <param name="vm-password" value="5000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="5000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 5000"/> <variable name="effective_caller_id_number" value="5000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> After 1000 calls, it is giving the same error -ERR DESTINATION_OUT_OF_ORDER Yes, my CPU (quad-core AMD) and RAM (2GB only) are under heavy stress by the time 1000 calls are being done, but is it not a little strange that every time the resources get exhausted at precisely 1000 calls! regards, Anita

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IVR (Calling card application) development
by Vitaly Nikolaev 09 Jan '12

09 Jan '12

Hello, I am choosing between ESL and mod_erlang, I will have around 1k simultaneous calls with 10 min average duration. I will definitely write it in erlang, but question is - how supported is mod_erlang ? Is anyone currently use it in production for heavy load application ? Thank you Vitaly

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Invalid UTF-8 character to ampersand, skip it
by Rodney 09 Jan '12

09 Jan '12

this is doing the same for me but only when they press my option for the Conference Call Count extension. did have any problems with the november msi (english windows version) still hoping it gets fixed. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, January 7, 2012 12:41 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 62 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Invalid UTF-8 character to ampersand, skip it (Juan Antonio Iba?ez Santorum) 2. Out of event dispatch threads! Slowing things down. (Stephen Wilde) 3. Need help using api_hangup_hook and session_in_hangup_hook (Herman Griffin) 4. 407 undefined? sip port 5060 (Rodney) Try to update to last git. There are some not supported chars at german language files. Regards 2012/1/7 Ryan How <rhow(a)exemail.com.au> Hi, > >I've just installed freeswitch on windows using the binary install >(FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). >Whenever a call is made the log fills up with about 42 "Invalid UTF-8 >character to ampersand, skip it". I haven't changed any config from the >default. Any pointers ? > >Thanks! > >Ryan > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > Hi, my system is showing in the log the following critical errors: 2012-01-07 18:13:34.978566 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:35.981506 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:36.975496 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:37.974385 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:38.977286 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:39.983208 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:40.999186 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:42.003136 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:27.210316 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:28.214168 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:29.211205 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:14:30.208196 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:31.210143 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break The CPU is running at 40%-50% of load and the number of sessions in below the limit, the same for sessions per seconds. There is any parameter that I have to adjust? StephenHello, I'm using the api_hangup_hook and session_in_hangup variables with a pyrun. How do I access the session object from inside the python module? --------------------------------- Here are some logs : Dialplan: sofia/external/3105795721(a)72.37.252.18 Absolute Condition [emergency] Dialplan: sofia/external/3105795721(a)72.37.252.18 Action set(session_in_hangup_hook=true) Dialplan: sofia/external/3105795721(a)72.37.252.18 Action set(api_hangup_hook=pyrun emergency.hangup) Dialplan: sofia/external/3105795721(a)72.37.252.18 Absolute Condition [emergency] . . 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:281 Call python script 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:284 Finished calling python script 2012-01-07 09:21:57.693488 [ERR] mod_python.c:293 Error calling python script 2012-01-07 09:21:57.693488 [ERR] mod_python.c:164 Python Error by calling script "emergency.hangup": <type 'exceptions.NameError'> Message: global name 'session' is not defined Exception: None Traceback (most recent call last) File: "/usr/local/freeswitch/scripts/python/emergency/hangup.py", line 34, in runtime ---------------------------------------------------------- Here is the simple script python module emergency.hangup: from freeswitch import * def runtime(arg): consoleLog("info", print(dir(session))) Thanks, Herman Vitalie, I asked my provider to switch up to port 5080 and everything is negotiating perfectly. thanks for your help. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, January 6, 2012 11:54 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 58 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. 407 undefined? sip port 5060 (Rodney) 2. Why database is locked? (Valery Kalinin) 3. Re: Why database is locked? (Ken Rice) 4. Re: 407 undefined? sip port 5060 (Vitalie Colosov) I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response: 2012-01-06 21:52:46.0 000000000 614??????? 6174530953 ( DID-IN:RNK) (H1 usinteractive-nextdi) 407 undefined I have the following gateway setup in C:\Program Files\FreeSWITCH\conf\sip_profiles\external <include> <gateway name="RNK_Test"> <param name="username" value="superadmin"/> <param name="password" value="fusionpbx"/> <param name="proxy" value="76.181.53.231:5060"/> <param name="expire-seconds" value="800"/> <param name="register" value="false"/> <param name="retry-seconds" value="60"/> <param name="context" value="default"/> </gateway> </include> It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. thank you. > Are you running the latest version of FS? FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > Also are you doing anything externally to mess with the sql db files? No. > Another possibility is a slow filesystem, we used to have people get this problem on suse linux. CentOS release 5.5 (Final) > You could try running your db dir in a ramdisk. These problems appeared after the upgrade version of FS. Thank you in advance... Re: [Freeswitch-users] Why database is locked? Blow away all the files in freeswitch/db and restart freeswitch Save the voicemail.db file if you don’t wanna lose all the voicemails and give that a try. That will force a reset of the all the databases and should clear any errors... On 1/6/12 10:00 PM, "Valery Kalinin" <valery.kalinin(a)gmail.com> wrote: > Are you running the latest version of FS? > >FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > >> Also are you doing anything externally to mess with the sql db files? > >No. > >> Another possibility is a slow filesystem, we used to have people get this >problem on suse linux. > >CentOS release 5.5 (Final) > >> You could try running your db dir in a ramdisk. > >These problems appeared after the upgrade version of FS. > > >Thank you in advance... > >>________________________________ >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > Ask your provider to send sip packets to 5080, instead of 5060. Every good provider should allow you to select the port YOU want, and not THEY want. If it will not help, then you need to make external profile listen to 5060, instead of 5080. So, you will be sending and receiving sip messages between you and provider, using port 5060 (which is usually used only for your internal profile and your clients). To change this, you need to modify conf/sip_profiles/external.xml <param name="sip-port" value="5080"/> replace to "5060" In this case, you will have to reconfigure internal profile to be on a different port, say 5062, and ask your clients to add :5062 to the sip server address when they configure their softphone/sip adapter. Maybe another option would be to consider provider as one of your internal clients, and allow password-less communication from his IP, but this is way too crazy to consider as a good option. Vitalie 2012/1/6 Rodney <notlikeme75(a)yahoo.com> I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response: > > >2012-01-06 21:52:46.0 000000000 614??????? 6174530953 ( DID-IN:RNK) (H1 usinteractive-nextdi) 407 undefined > > > > > > > >I have the following gateway setup in C:\Program Files\FreeSWITCH\conf\sip_profiles\external > > > > ><include> > <gateway name="RNK_Test"> > <param name="username" value="superadmin"/> > <param name="password" value="fusionpbx"/> > <param name="proxy" value="76.181.53.231:5060"/> > <param name="expire-seconds" value="800"/> > <param name="register" value="false"/> > <param name="retry-seconds" value="60"/> > <param name="context" value="default"/> > </gateway> ></include> > > >It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. > > >thank you. > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > _______________________________________________ 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 _______________________________________________ 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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