[Freeswitch-users] Help! No output to CLI with console_log() from script
I'm trying to get output to my CLI in Windows XP when running this script by
dialing 8337 but the output Hello World! doesn't show up. The rest works fine.
What's wrong with console_log(Hello World!\n) ? Where has the output gone?
console_log(Hello World!\n);
var languageCode = en;
var soundDir = sound/;
function playFile(fileName, callBack, callBackArgs)
{
??? session.streamFile(soundDir + languageCode +? / + fileName);
}
session.answer();
playFile(HelloWorld.wav);
exit();
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Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?
On Tue, Oct 28, 2008 at 10:26 PM, Ryan McDougall [EMAIL PROTECTED] wrote: Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers, As my research continues I see how badly phrased my question was. A stateful SIP proxy and a B2BUA are very different beasts it would seem. The former is best accomplished by taking Kamailio and using its custom scripting language to write into to a database using one of its DB plugins. The latter can only be accomplished by using FreeSwitch, since the above scripting language is rather limited. My revised question is this: What API does does FreeSwitch expose to implement a SIP B2BUA, and where? In other words can you point me to any documentation for implementing a SIP B2BUA? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@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] System requirements
Hello What kind of system requirements do FS have to handle about 500 simultaneous calls? The specsheet in the wiki have Minimum/Recommended System Requirements, but they just talk about memory and disk space (and doesnt mention how many calls the recomendations are for). Thanks, Jonas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?
FreeSwitch *IS* a b2bua... You don't have to implement anything... There are already plenty of high level (and config based) options to set up a b2bua in a variety of configurations... Check out the wiki (http://wiki.freeswitch.org) for some example configs From: Ryan McDougall [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 09:31:59 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent? On Tue, Oct 28, 2008 at 10:26 PM, Ryan McDougall [EMAIL PROTECTED] wrote: Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers, As my research continues I see how badly phrased my question was. A stateful SIP proxy and a B2BUA are very different beasts it would seem. The former is best accomplished by taking Kamailio and using its custom scripting language to write into to a database using one of its DB plugins. The latter can only be accomplished by using FreeSwitch, since the above scripting language is rather limited. My revised question is this: What API does does FreeSwitch expose to implement a SIP B2BUA, and where? In other words can you point me to any documentation for implementing a SIP B2BUA? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?
Maybe I should be more specific, and you can tell me where I am going wrong. I wish to implement SIP-based IM, presence, and voice in a MMO game. I need to take incoming SIP requests and authorize them with my own UDP or XMLRPC-based authentication server, so I can sync SIP and game authorizations. What do you suggest? Cheers, On Wed, Oct 29, 2008 at 9:38 AM, Ken Rice [EMAIL PROTECTED] wrote: FreeSwitch *IS* a b2bua... You don't have to implement anything... There are already plenty of high level (and config based) options to set up a b2bua in a variety of configurations... Check out the wiki (http://wiki.freeswitch.org) for some example configs From: Ryan McDougall [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 09:31:59 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent? On Tue, Oct 28, 2008 at 10:26 PM, Ryan McDougall [EMAIL PROTECTED] wrote: Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers, As my research continues I see how badly phrased my question was. A stateful SIP proxy and a B2BUA are very different beasts it would seem. The former is best accomplished by taking Kamailio and using its custom scripting language to write into to a database using one of its DB plugins. The latter can only be accomplished by using FreeSwitch, since the above scripting language is rather limited. My revised question is this: What API does does FreeSwitch expose to implement a SIP B2BUA, and where? In other words can you point me to any documentation for implementing a SIP B2BUA? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem with mod_openzap.c and SS7boost
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Anthony, ok, here the console output. There were no purple lines 2008-10-29 09:04:57 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/softswitch_side/[EMAIL PROTECTED] Execute bridge(OpenZAP/1/a/44180002850) 2008-10-29 09:04:57 [DEBUG] ozmod_ss7_boost.c:225 ss7_boost_channel_request() TX EVENT: CALL_START:(80) [w1g1] CSid=2 Seq=1 Cn=[hk] Cd=[44180002850] Ci=[31152112850] 2008-10-29 09:04:58 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/softswitch_side/[EMAIL PROTECTED] entering state [terminated] 2008-10-29 09:04:58 [NOTICE] sofia.c:2765 sofia_handle_sip_i_state() Hangup sofia/softswitch_side/[EMAIL PROTECTED] [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-10-29 09:04:58 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal sofia/softswitch_side/[EMAIL PROTECTED] [KILL] 2008-10-29 09:04:58 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal sofia/softswitch_side/[EMAIL PROTECTED] [BREAK] 2008-10-29 09:04:58 [DEBUG] ozmod_ss7_boost.c:933 zap_ss7_boost_run() RX EVENT (N): CALL_START_ACK:(81) [w1g30] Rc=0 CSid=2 Seq=3 2008-10-29 09:04:58 [DEBUG] ss7_boost_client.c:300 __ss7bc_connection_read() Rx sync ok 2008-10-29 09:04:58 [DEBUG] ozmod_ss7_boost.c:297 handle_call_start_ack() Assign chan 1:29 (1:30) CSid=2 2008-10-29 09:04:58 [CRIT] ozmod_ss7_boost.c:245 ss7_boost_channel_request() setting init state to progress_media 2008-10-29 09:04:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMA 20ms 2008-10-29 09:04:58 [DEBUG] mod_openzap.c:1022 channel_outgoing_channel() Connect outbound channel OpenZAP/1:29/44180002850 2008-10-29 09:04:58 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel OpenZAP/1:29/44180002850 [a231e1ef-2cfc-4de2-ab73-2175c184dff5] 2008-10-29 09:04:58 [DEBUG] mod_openzap.c:1031 channel_outgoing_channel() (OpenZAP/1:29/44180002850) State Change CS_NEW - CS_INIT 2008-10-29 09:04:58 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal OpenZAP/1:29/44180002850 [BREAK] 2008-10-29 09:04:58 [DEBUG] switch_ivr_originate.c:1598 switch_ivr_originate() Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2008-10-29 09:04:58 [NOTICE] switch_ivr_originate.c:1637 switch_ivr_originate() Hangup OpenZAP/1:29/44180002850 [CS_INIT] [ORIGINATOR_CANCEL] 2008-10-29 09:04:58 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal OpenZAP/1:29/44180002850 [KILL] 2008-10-29 09:04:58 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal OpenZAP/1:29/44180002850 [BREAK] 2008-10-29 09:04:58 [INFO] mod_dptools.c:1843 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/softswitch_side/[EMAIL PROTECTED]) State EXECUTE going to sleep 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (sofia/softswitch_side/[EMAIL PROTECTED]) Running State Change CS_HANGUP 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/softswitch_side/[EMAIL PROTECTED]) State HANGUP 2008-10-29 09:04:58 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel sofia/softswitch_side/[EMAIL PROTECTED] hanging up, cause: ORIGINATOR_CANCEL 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/softswitch_side/[EMAIL PROTECTED] Standard HANGUP, cause: ORIGINATOR_CANCEL 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/softswitch_side/[EMAIL PROTECTED]) State HANGUP going to sleep 2008-10-29 09:04:58 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 3 (sofia/softswitch_side/[EMAIL PROTECTED]) Locked, Waiting on external entities 2008-10-29 09:04:58 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 3 (sofia/softswitch_side/[EMAIL PROTECTED]) Ended 2008-10-29 09:04:58 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/softswitch_side/[EMAIL PROTECTED] [CS_HANGUP] 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (OpenZAP/1:29/44180002850) Running State Change CS_HANGUP 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (OpenZAP/1:29/44180002850) State HANGUP 2008-10-29 09:04:58 [DEBUG] mod_openzap.c:476 channel_on_hangup() OpenZAP/1:29/44180002850 CHANNEL HANGUP 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:29/44180002850 Standard HANGUP, cause: ORIGINATOR_CANCEL 2008-10-29 09:04:58 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (OpenZAP/1:29/44180002850) State HANGUP going to sleep 2008-10-29 09:04:58 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 4 (OpenZAP/1:29/44180002850) Locked, Waiting
Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?
In that case you should look at the xml-curl stuff on the wiki... With it you can feed the user auth information from an external database From: Ryan McDougall [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 09:47:03 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent? Maybe I should be more specific, and you can tell me where I am going wrong. I wish to implement SIP-based IM, presence, and voice in a MMO game. I need to take incoming SIP requests and authorize them with my own UDP or XMLRPC-based authentication server, so I can sync SIP and game authorizations. What do you suggest? Cheers, On Wed, Oct 29, 2008 at 9:38 AM, Ken Rice [EMAIL PROTECTED] wrote: FreeSwitch *IS* a b2bua... You don't have to implement anything... There are already plenty of high level (and config based) options to set up a b2bua in a variety of configurations... Check out the wiki (http://wiki.freeswitch.org) for some example configs From: Ryan McDougall [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 09:31:59 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent? On Tue, Oct 28, 2008 at 10:26 PM, Ryan McDougall [EMAIL PROTECTED] wrote: Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers, As my research continues I see how badly phrased my question was. A stateful SIP proxy and a B2BUA are very different beasts it would seem. The former is best accomplished by taking Kamailio and using its custom scripting language to write into to a database using one of its DB plugins. The latter can only be accomplished by using FreeSwitch, since the above scripting language is rather limited. My revised question is this: What API does does FreeSwitch expose to implement a SIP B2BUA, and where? In other words can you point me to any documentation for implementing a SIP B2BUA? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help! No output to CLI with console_log() from script
I changed console_log(Hello World!\n) to console_log(debug, Hello
World!\n) and that didn't work either. Finally, I got console_log(notice,
Hello World!\n) to output to the CLI.
Where is the output with debug going?
-Original Message-
From: [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 28 Oct 2008 11:14 pm
Subject: [Freeswitch-users] Help! No output to CLI with console_log() from
script
I'm trying to get output to my CLI in Windows XP when running this script by
dialing 8337 but the output Hello World! doesn't show up. The rest works fine.
What's wrong with console_log(Hello World!\n) ? Where has the output gone?
console_log(Hello World!\n);
var languageCode = en;
var soundDir = sound/;
function playFile(fileName, callBack, callBackArgs)
{
??? session.streamFile(soundDir + languageCode +? / + fileName);
}
session.answer();
playFile(HelloWorld.wav);
exit();
McCain or Obama? Stay updated on coverage of the Presidential race while you
browse - Download Now!
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Re: [Freeswitch-users] Help! No output to CLI with console_log() from script
Thanks for the pointer. The fsctl loglevel 7 didn't seem to work but console
loglevel 7 did.
-Original Message-
From: Matt Klein [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 29 Oct 2008 1:28 am
Subject: Re: [Freeswitch-users] Help! No output to CLI with console_log() from
script
You can enter fsctl loglevel 7 for debug output in the CLI.
It looks like your configuration for the loglevel of CLI output is set too low.
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Increase_Debug_Output
m
On Wed, 29 Oct 2008, [EMAIL PROTECTED] wrote:
I changed console_log(Hello World!\n) to console_log(debug, Hello
World!\n) and that didn't work either. Finally, I got console_log(notice,
Hello World!\n) to output to the CLI.
Where is the output with debug going?
-Original Message-
From: [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 28 Oct 2008 11:14 pm
Subject: [Freeswitch-users] Help! No output to CLI with console_log() from
script
I'm trying to get output to my CLI in Windows XP when running this script by
dialing 8337 but the output Hello World! doesn't show up. The rest works fine.
What's wrong with console_log(Hello World!\n) ? Where has the output gone?
console_log(Hello World!\n);
var languageCode = en;
var soundDir = sound/;
function playFile(fileName, callBack, callBackArgs)
{
??? session.streamFile(soundDir + languageCode +? / + fileName);
}
session.answer();
playFile(HelloWorld.wav);
exit();
McCain or Obama? Stay updated on coverage of the Presidential race while you
browse - Download Now!
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Re: [Freeswitch-users] What happend to variable_* in socket_outbound?
they are still there, at least if you register to myevents or all events. i use the default settings from fs and get plenty of them. 2008/10/28 Andy Spitzer [EMAIL PROTECTED]: Woof! I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back? --Woof! ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] System requirements
On Wed, Oct 29, 2008 at 1:03 PM, Jonas Gauffin [EMAIL PROTECTED]wrote: Hello What kind of system requirements do FS have to handle about 500 simultaneous calls? The specsheet in the wiki have Minimum/Recommended System Requirements, but they just talk about memory and disk space (and doesnt mention how many calls the recomendations are for). Look at FAQ ram Q: How many concurrent calls can it support? Any benchmarks? - FreeSWITCH has done 3000 concurrent channels with media on a dual woodcrest 2.0 GHz (1 year old). The test was to play back a wav file so there was slight transcoding going on. - With current modern cpus you should be able to do 6000 channels per chassis, both because there are 2x the cores as well as 50% additional cycles per core. A dual clovertown (quad instead of dual core) at 3.0GHz should be able to at least double your channel capacity ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP incoming call routing
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gateway ram ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP incoming call routing
Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register. -Saurabh Date: Wed, 29 Oct 2008 15:12:28 +0530From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP incoming call routing On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gateway ram _ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] System requirements
doh. It was too close :) On Wed, Oct 29, 2008 at 10:40 AM, ram [EMAIL PROTECTED] wrote: On Wed, Oct 29, 2008 at 1:03 PM, Jonas Gauffin [EMAIL PROTECTED]wrote: Hello What kind of system requirements do FS have to handle about 500 simultaneous calls? The specsheet in the wiki have Minimum/Recommended System Requirements, but they just talk about memory and disk space (and doesnt mention how many calls the recomendations are for). Look at FAQ ram Q: How many concurrent calls can it support? Any benchmarks? - FreeSWITCH has done 3000 concurrent channels with media on a dual woodcrest 2.0 GHz (1 year old). The test was to play back a wav file so there was slight transcoding going on. - With current modern cpus you should be able to do 6000 channels per chassis, both because there are 2x the cores as well as 50% additional cycles per core. A dual clovertown (quad instead of dual core) at 3.0GHz should be able to at least double your channel capacity ___ Freeswitch-users mailing list Freeswitch-users@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@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] SIP incoming call routing
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh _ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Cepstral 5.1 no sound
I reverted to this old libs/apr and after compiling the complete freeswitch (compiling only apr did't work) it finally worked. Thanks for your support Peter Wasim Baig schrieb: On Wed, Oct 29, 2008 at 7:20 AM, Peter P GMX [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I did a svn log: /usr/src/freeswitch/libs/apr# svn log r9605 | mikej | 2008-09-20 02:05:00 +0200 (Sa, 20 Sep 2008) | 1 line hack for now until we ditch apr dso code completely probably just reverting to remove this should be fine -wasim r8754 | anthm | 2008-06-04 19:53:33 +0200 (Mi, 04 Jun 2008) | 1 line prevent endless loop r6626 | mikej | 2007-12-11 10:52:42 +0100 (Di, 11 Dez 2007) | 1 line . . . Which one do I have to revert to? Best regards Peter Anthony Minessale schrieb: There is something that a patch to apr we did broke and we are working on it still. Do svn log on libs/apr and revert the last patch for a temp fix On 10/28/08, Michael Collins [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It would seem you are ahead of me... sorry I couldn't be of further assistance. -MC -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:freeswitch- mailto:freeswitch- [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Peter P GMX Sent: Tuesday, October 28, 2008 4:36 PM To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Cepstral 5.1 no sound I have only German sound files at present. So I symlinked them also to libceplang_en.so* libceplex_en.so* and did an ldconfig again. Here is a list of libs in /opt/swift/lib lrwxrwxrwx 1 root root 20 2008-07-17 18:10 libceplang_de.so - libceplang_de.so.5.1 lrwxrwxrwx 1 root root 20 2008-07-17 18:10 libceplang_de.so.5 - libceplang_de.so.5.1 -rwxrwxr-x 1 root root 170379 2008-07-08 21:19 libceplang_de.so.5.1 lrwxrwxrwx 1 root root 20 2008-10-28 23:25 libceplang_en.so - libceplang_de.so.5.1 lrwxrwxrwx 1 root root 20 2008-10-28 23:25 libceplang_en.so.5 - libceplang_de.so.5.1 lrwxrwxrwx 1 root root 20 2008-10-28 23:25 libceplang_en.so.5.1 - libceplang_de.so.5.1 lrwxrwxrwx 1 root root 19 2008-07-17 18:10 libceplex_de.so - libceplex_de.so.5.1 lrwxrwxrwx 1 root root 19 2008-07-17 18:10 libceplex_de.so.5 - libceplex_de.so.5.1 -rwxrwxr-x 1 root root 614294 2008-07-08 21:19 libceplex_de.so.5.1 lrwxrwxrwx 1 root root 19 2008-10-28 23:26 libceplex_en.so - libceplex_de.so.5.1 lrwxrwxrwx 1 root root 19 2008-10-28 23:26 libceplex_en.so.5 - libceplex_de.so.5.1 lrwxrwxrwx 1 root root 15 2008-07-17 18:10 libswift.so - libswift.so.5.1 lrwxrwxrwx 1 root root 15 2008-07-17 18:10 libswift.so.5 - libswift.so.5.1 -rwxrwxr-x 1 root root 1724351 2008-07-08 21:19 libswift.so.5.1 This at least didn't change anything. Best regards Peter Michael Collins schrieb: Bummer. I had some issues with the 5.0 version but that was because of file naming issues. I finally created symlinks and got it working. I've not tried 5.1 yet. When I get a minute I will and I'll see if I get the same error or not. For kicks, can you try a different language? I'm just curious to see what would happen. -MC -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:freeswitch- mailto:freeswitch- [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Peter P GMX Sent: Tuesday, October 28, 2008 4:16 PM To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Cepstral 5.1 no sound Hello Michael, No, I startet with a 5.1 installation. Cepstral works on the command line opt/swift/bin/swift -o hello.wav 'Hallo Peter' And the voice is registered: [EMAIL PROTECTED]:/opt/swift/bin# ./swift --voices Swift command-line synthesis program Version 5.1.0 of July 2008 Copyright (c) 2000-2006, Cepstral LLC. Voice | Version | Lic? | Gender | Age | Language | Sample Rate
Re: [Freeswitch-users] Help! No output to CLI with console_log() from script
fsctl loglevel is the global loglevel meaning when you change it it will
effect all logger modules it's a system level command.
console loglevel is mod_console's log level. mod_console is a logger module
that sits on the console and filters which lines will
print on the screen based on level and some string matching params found in
its config.
On Wed, Oct 29, 2008 at 3:57 AM, [EMAIL PROTECTED] wrote:
Thanks for the pointer. The fsctl loglevel 7 didn't seem to work but
console loglevel 7 did.
-Original Message-
From: Matt Klein [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 29 Oct 2008 1:28 am
Subject: Re: [Freeswitch-users] Help! No output to CLI with console_log()
from script
You can enter fsctl loglevel 7 for debug output in the CLI.
It looks like your configuration for the loglevel of CLI output is set too
low.
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Increase_Debug_Output
m
On Wed, 29 Oct 2008, [EMAIL PROTECTED] wrote:
I changed console_log(Hello World!\n) to console_log(debug, Hello
World!\n) and that didn't work either. Finally, I got console_log(notice,
Hello World!\n) to output to the CLI.
Where is the output with debug going?
-Original Message-
From: [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 28 Oct 2008 11:14 pm
Subject: [Freeswitch-users] Help! No output to CLI with console_log() from
script
I'm trying to get output to my CLI in Windows XP when running this script by
dialing 8337 but the output Hello World! doesn't show up. The rest works
fine.
What's wrong with console_log(Hello World!\n) ? Where has the output gone?
console_log(Hello World!\n);
var languageCode = en;
var soundDir = sound/;
function playFile(fileName, callBack, callBackArgs)
{
??? session.streamFile(soundDir + languageCode +? / + fileName);
}
session.answer();
playFile(HelloWorld.wav);
exit();
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Re: [Freeswitch-users] What happend to variable_* in socket_outbound?
by default only some hand-picked events have all the variables due to people complaining that they had too much info ;) if you want every event to have them you can execute verbose_events app in your dialplan right before you call socket On Wed, Oct 29, 2008 at 4:03 AM, Dennis [EMAIL PROTECTED] wrote: they are still there, at least if you register to myevents or all events. i use the default settings from fs and get plenty of them. 2008/10/28 Andy Spitzer [EMAIL PROTECTED]: Woof! I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back? --Woof! ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?
cool! is it fun? On Wed, Oct 29, 2008 at 2:47 AM, Ryan McDougall [EMAIL PROTECTED] wrote: Maybe I should be more specific, and you can tell me where I am going wrong. I wish to implement SIP-based IM, presence, and voice in a MMO game. I need to take incoming SIP requests and authorize them with my own UDP or XMLRPC-based authentication server, so I can sync SIP and game authorizations. What do you suggest? Cheers, On Wed, Oct 29, 2008 at 9:38 AM, Ken Rice [EMAIL PROTECTED] wrote: FreeSwitch *IS* a b2bua... You don't have to implement anything... There are already plenty of high level (and config based) options to set up a b2bua in a variety of configurations... Check out the wiki (http://wiki.freeswitch.org) for some example configs From: Ryan McDougall [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 09:31:59 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent? On Tue, Oct 28, 2008 at 10:26 PM, Ryan McDougall [EMAIL PROTECTED] wrote: Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers, As my research continues I see how badly phrased my question was. A stateful SIP proxy and a B2BUA are very different beasts it would seem. The former is best accomplished by taking Kamailio and using its custom scripting language to write into to a database using one of its DB plugins. The latter can only be accomplished by using FreeSwitch, since the above scripting language is rather limited. My revised question is this: What API does does FreeSwitch expose to implement a SIP B2BUA, and where? In other words can you point me to any documentation for implementing a SIP B2BUA? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Authorizing Anonynous Devices
all this to avoid just making another profile on a different port that has
inbound calls sandboxed into a special public context?
if you add the port to your srv records nobody would even know.
On Tue, Oct 28, 2008 at 3:02 PM, Klaus Teller [EMAIL PROTECTED] wrote:
Hi Folks,
I need some additional help with this issue. I already had some from Brian
i'm but still not able to move forward.
I want a non-registered device to be able to call extension 56900 in my
Freeswitch in such a way that i can manage the call using the socket
interface. I believe the issue i'm having is that of permissions.
Based on what was suggested sofar, here is what i did.
1) Added following extension in dialplan/default.xml
extension name=check_auth continue=true
condition field=${sip_authorized} expression=^true$
break=never
anti-action application=respond data=407/
/condition
/extension
2) Set auth-calls to false in internal.xml. That is
param name=auth-calls value=$${internal_auth_calls}/
was replaced with:
param name=auth-calls value=false/
3) Changed acl.com.xml by replacing
list name=domains default=DENY
node type=allow domain=$${domain}/
/list
with
list name=domains default=allow
node type=allow domain=$${domain}/
/list
Now here is the result i get after these changes:
a) Anonymous non-registered device can call registered soft phone at
extension 1003
b) Anonymous non-registered device cannot call 56900 that needs to be
managed via socket interface (error message 480). Also 9000 cannot be
called.
c) Registered soft phone (extension 1003) cannot call 56900
d) Registered soft phone (ext 1003) can call registered soft phone (ext
1000).
If i perform only step 1 and 3 (i.e. auth-calls not set to false), a)
become impossible, b) remains wrong, c) is now possible (i.e. socket
interface being notified about call at 56900), while d) remains. valid.
Disabling any of 1) or 3) would result into calls by non-registered device
being rejected.
Any idea what else can be tried?
Thanks,
Klaus.
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Re: [Freeswitch-users] Authorizing Anonynous Devices
Sorry if i gave the impression i'm tried to avoid something. There is nothing
i'm trying to avoid, i'm just ignorant.
So how can i translate your recommendation into practice? What parameters do i
need to set/change?
Thanks,
Klaus.
Original-Nachricht
Datum: Wed, 29 Oct 2008 08:13:58 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Authorizing Anonynous Devices
all this to avoid just making another profile on a different port that has
inbound calls sandboxed into a special public context?
if you add the port to your srv records nobody would even know.
On Tue, Oct 28, 2008 at 3:02 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi Folks,
I need some additional help with this issue. I already had some from
Brian
i'm but still not able to move forward.
I want a non-registered device to be able to call extension 56900 in my
Freeswitch in such a way that i can manage the call using the socket
interface. I believe the issue i'm having is that of permissions.
Based on what was suggested sofar, here is what i did.
1) Added following extension in dialplan/default.xml
extension name=check_auth continue=true
condition field=${sip_authorized} expression=^true$
break=never
anti-action application=respond data=407/
/condition
/extension
2) Set auth-calls to false in internal.xml. That is
param name=auth-calls value=$${internal_auth_calls}/
was replaced with:
param name=auth-calls value=false/
3) Changed acl.com.xml by replacing
list name=domains default=DENY
node type=allow domain=$${domain}/
/list
with
list name=domains default=allow
node type=allow domain=$${domain}/
/list
Now here is the result i get after these changes:
a) Anonymous non-registered device can call registered soft phone at
extension 1003
b) Anonymous non-registered device cannot call 56900 that needs to be
managed via socket interface (error message 480). Also 9000 cannot be
called.
c) Registered soft phone (extension 1003) cannot call 56900
d) Registered soft phone (ext 1003) can call registered soft phone (ext
1000).
If i perform only step 1 and 3 (i.e. auth-calls not set to false), a)
become impossible, b) remains wrong, c) is now possible (i.e. socket
interface being notified about call at 56900), while d) remains. valid.
Disabling any of 1) or 3) would result into calls by non-registered
device
being rejected.
Any idea what else can be tried?
Thanks,
Klaus.
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[Freeswitch-users] mod_cdr revival (or new module maybe)
Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Unsure at this time. There has been some work on mod_cdr_odbc. We generally advise against direct to db cdr methods without a very robust backup method for when the db is down. On Oct 29, 2008, at 9:57 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?
Making the game part is fun! Not understanding telecoms or pbxs isn't. ;) I'll do some more experiments and get back to you guys. Thanks for the help so far. Cheers, On Wed, Oct 29, 2008 at 10:11 PM, Anthony Minessale [EMAIL PROTECTED] wrote: cool! is it fun? On Wed, Oct 29, 2008 at 2:47 AM, Ryan McDougall [EMAIL PROTECTED] wrote: Maybe I should be more specific, and you can tell me where I am going wrong. I wish to implement SIP-based IM, presence, and voice in a MMO game. I need to take incoming SIP requests and authorize them with my own UDP or XMLRPC-based authentication server, so I can sync SIP and game authorizations. What do you suggest? Cheers, On Wed, Oct 29, 2008 at 9:38 AM, Ken Rice [EMAIL PROTECTED] wrote: FreeSwitch *IS* a b2bua... You don't have to implement anything... There are already plenty of high level (and config based) options to set up a b2bua in a variety of configurations... Check out the wiki (http://wiki.freeswitch.org) for some example configs From: Ryan McDougall [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 09:31:59 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent? On Tue, Oct 28, 2008 at 10:26 PM, Ryan McDougall [EMAIL PROTECTED] wrote: Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers, As my research continues I see how badly phrased my question was. A stateful SIP proxy and a B2BUA are very different beasts it would seem. The former is best accomplished by taking Kamailio and using its custom scripting language to write into to a database using one of its DB plugins. The latter can only be accomplished by using FreeSwitch, since the above scripting language is rather limited. My revised question is this: What API does does FreeSwitch expose to implement a SIP B2BUA, and where? In other words can you point me to any documentation for implementing a SIP B2BUA? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help! No output to CLI with console_log() from script
Hi Anthony,
Thanks for pointing out the distinction between the two.
I don't need to use debug now since any of the other levels work (info,
notice, warning, err, crit, alert) but probably will later. What is
the default setting? I changed it to loglevel 6 afterwards.
?In any case, I not getting output with console_log() having no level
specified??
Also, could you look at this post. The very last lines indicate a problem with
hearing a lot of static on my .wav file. The static isn't there on any of the
standard (non-FreeSWITCH) players.
The stuff prior is just my perspective as a newbie to FreeSWITCH (amoung other
things) on the JavaScript Quickstart.
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05193.html
Cheers.
-Original Message-
From: Anthony Minessale [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 29 Oct 2008 6:07 am
Subject: Re: [Freeswitch-users] Help! No output to CLI with console_log() from
script
fsctl loglevel is the global loglevel meaning when you change it it will effect
all logger modules it's a system level command.
console loglevel is mod_console's log level.? mod_console is a logger module
that sits on the console and filters which lines will
print on the screen based on level and some string matching params found in its
config.
On Wed, Oct 29, 2008 at 3:57 AM, [EMAIL PROTECTED] wrote:
Thanks for the pointer. The fsctl loglevel 7 didn't seem to work but console
loglevel 7 did.
-Original Message-
From: Matt Klein [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 29 Oct 2008 1:28 am
Subject: Re: [Freeswitch-users] Help! No output to CLI with console_log() from
script
You can enter fsctl loglevel 7 for debug output in the CLI.
It looks like your configuration for the loglevel of CLI output is set too low.
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Increase_Debug_Output
m
On Wed, 29 Oct 2008, [EMAIL PROTECTED] wrote:
I changed console_log(Hello World!\n) to console_log(debug, Hello
World!\n) and that didn't work either. Finally, I got console_log(notice,
Hello World!\n) to output to the CLI.
Where is the output with debug going?
-Original Message-
From: [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 28 Oct 2008 11:14 pm
Subject: [Freeswitch-users] Help! No output to CLI with console_log() from
script
I'm trying to get output to my CLI in Windows XP when running this script by
dialing 8337 but the output Hello World! doesn't show up. The rest works fine.
What's wrong with console_log(Hello World!\n) ? Where has the output gone?
console_log(Hello World!\n);
var languageCode = en;
var soundDir = sound/;
function playFile(fileName, callBack, callBackArgs)
{
??? session.streamFile(soundDir + languageCode +? / + fileName);
}
session.answer();
playFile(HelloWorld.wav);
exit();
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Re: [Freeswitch-users] Authorizing Anonynous Devices
oh that's good news!
sorry, i assumed you must have had a reason to do it that way.
just create another profile that uses a another port and a public context
with very little extensions in it and no auth options.
then in that public context you can make extens to transfer the call into
your real dialplan when they are appropriate.
the defaults out of the box have something like this in place already iirc.
On Wed, Oct 29, 2008 at 8:38 AM, Klaus Teller [EMAIL PROTECTED] wrote:
Sorry if i gave the impression i'm tried to avoid something. There is
nothing i'm trying to avoid, i'm just ignorant.
So how can i translate your recommendation into practice? What parameters
do i need to set/change?
Thanks,
Klaus.
Original-Nachricht
Datum: Wed, 29 Oct 2008 08:13:58 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Authorizing Anonynous Devices
all this to avoid just making another profile on a different port that
has
inbound calls sandboxed into a special public context?
if you add the port to your srv records nobody would even know.
On Tue, Oct 28, 2008 at 3:02 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi Folks,
I need some additional help with this issue. I already had some from
Brian
i'm but still not able to move forward.
I want a non-registered device to be able to call extension 56900 in my
Freeswitch in such a way that i can manage the call using the socket
interface. I believe the issue i'm having is that of permissions.
Based on what was suggested sofar, here is what i did.
1) Added following extension in dialplan/default.xml
extension name=check_auth continue=true
condition field=${sip_authorized} expression=^true$
break=never
anti-action application=respond data=407/
/condition
/extension
2) Set auth-calls to false in internal.xml. That is
param name=auth-calls value=$${internal_auth_calls}/
was replaced with:
param name=auth-calls value=false/
3) Changed acl.com.xml by replacing
list name=domains default=DENY
node type=allow domain=$${domain}/
/list
with
list name=domains default=allow
node type=allow domain=$${domain}/
/list
Now here is the result i get after these changes:
a) Anonymous non-registered device can call registered soft phone at
extension 1003
b) Anonymous non-registered device cannot call 56900 that needs to be
managed via socket interface (error message 480). Also 9000 cannot be
called.
c) Registered soft phone (extension 1003) cannot call 56900
d) Registered soft phone (ext 1003) can call registered soft phone (ext
1000).
If i perform only step 1 and 3 (i.e. auth-calls not set to false), a)
become impossible, b) remains wrong, c) is now possible (i.e. socket
interface being notified about call at 56900), while d) remains. valid.
Disabling any of 1) or 3) would result into calls by non-registered
device
being rejected.
Any idea what else can be tried?
Thanks,
Klaus.
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Re: [Freeswitch-users] SIP incoming call routing
whatever you put in the extension param in the gateway should control what destination_number it has in the inbound call. you can also do your regex in your dialplan on any of the info in the sip packet besides destination number if you wish. On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register. -Saurabh -- Date: Wed, 29 Oct 2008 15:12:28 +0530 From: [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP incoming call routing On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gateway ram -- When your life is on the go—take your life with you. Try Windows Mobile(R) today http://clk.atdmt.com/MRT/go/115298558/direct/01/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dptools read command doesn't appear to work properly (how to read dtmf via event socket)
I tried to read dtmfs via event socket and came across this thread. What do you mean by: If you're using event socket you have really no reason to use the read application. Is there another chance to do this? Currently I do the following and this isn't successful: I send the following message to FS: SendMsg 4f286c18-a5de-11dd-bef3-4b1a61d55c50 call-command: execute execute-app-name: read execute-app-arg: 0 10 ivr/8000/ivr-enter_ext.wav dtmfdtmf 1 #,* event-lock:true I type 4 digits on the phone 1 2 3 4 plus # I receive 5 separate DTMF events DTMF-Digit: 1 DTMF-Digit: 2 DTMF-Digit: 3 DTMF-Digit: 4 DTMF-Digit: %23 The final event however doesn't deliver the parameter dtmfdtmf as defined above. So I cannot receive the dtmf digits this way. I grepped the network traffic to be sure I haven't missed anything. Any idea? Best regards Peter Brian West schrieb: If you're using event socket you have really no reason to use the read application. The way you invoked it would collect 0 to 4 digits into the variable 'digits'. /b On Sep 12, 2008, at 4:27 PM, Luke Graybill wrote: I am having a hard time getting this command to work properly. Here is the c output from my test session, where I dial in to ext 500 with my sip client (Ekiga): listening on localhost:8084 IPv4Address(TCP, '127.0.0.1 http://127.0.0.1', 36685): Connected 'connect' Caller: Killarny - 'myevents' result was: Events Enabled 'sendmsg\ncall-command: execute\nexecute-app-name: answer' OK 'sendmsg\ncall-command: execute\nexecute-app-name: read\nexecute-app-arg: 0 4 conference/8000/conf-pin.wav digits 1 #' OK DTMF: 1 DTMF: 2 DTMF: 3 DTMF: 4 DTMF: 5 DTMF: 6 DTMF: 7 DTMF: 8 DTMF: 9 DTMF: 0 DTMF: * DTMF: # IPv4Address(TCP, '127.0.0.1 http://127.0.0.1', 36685): Connection was closed cleanly. I couldn't figure out how to get read to function without passing a wav file (I plan to use TTS for the voice prompts, and I don't need a wav file to play here) so I just plugged in a random wav that comes with freeswitch for testing. As you can see, freeswitch reports that the command executed properly, illustrated by the OK (which was a reply-text: +OK) but the read application doesn't seem to be functioning beyond playing the given wav file (and truncating the first half of it as well). Pressing digits in the sip client results in DTMF events being sent to the socket. Note that the DTMF events stop at # because I manually shut down the connection, not because of anything that freeswitch does. Am I doing something wrong here? I feel like I'm following the available documentation given at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read but I am having no luck. Is the read application basically a dead command, and should I be instead just using DTMF event output? Any help that can be provided would be appreciated, thanks. Killarny ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
This all seems right and would make a great wiki page.
What you have described *should* work.
when a phone registers try doing
sofia_contact [EMAIL PROTECTED]
from the cli on each box and see what you get.
you can also use this function in the dialpan
${sofia_contact([EMAIL PROTECTED])}
check that they are both using the same domain name as the profile name
or at least have an alais for it etc.
if it's a bug i can fix it pretty fast as that is the intended behaviour
perhaps you can join irc and get us in the box(s) to have a look at it as we
do not have that situation labbed up anywhere.
On Tue, Oct 28, 2008 at 6:41 PM, Marc Lewis [EMAIL PROTECTED] wrote:
I am in the process of making my FreeSWITCH installation highly
available and I'm running into a couple of snags that was hoping that
someone may have some insight on.
First, the setup as it is now.
There are two installations of FS on two different servers, lets call
them fs1 and fs2. They each pull their configurations, dialplan,
directory and post CDR's all using mod_curl from a central web server.
That part works great.
Calls into and out of FS go through an OpenSER proxy set up using
carrierroute. That part also works great for outbound calls to the
PSTN. Inbound calls also come in through this OpenSER proxy and get
routed to the primary switch fs1. That also works perfectly as long as
its going to fs1.
fs1 and fs2 are both setup to use an ODBC connection to store
registrations. This is pointed to a MySQL database made highly
available using the RedHat Cluster Suite on a shared fibre channel
partition. fs1 and fs2 both share the same database. Voicemail storage
on fs1 is directly mounted on a GFS2 partition, fs2 is mounting the
shared storage from a different server via NFS for no single point of
failure.
For the phones, I have them setup to use SRV records and have fs1 at
priority 10 and fs2 at priority 20 for acme.domain.com. I've tested
this and phones register to the correct server and the sip_registration
table shows either fs1 or fs2 as the hostname as I would expect.
Here is the problem. If user [EMAIL PROTECTED] registers on fs2 and a
call comes in from the OpenSER proxy to fs1, bridging the call to
/sofia/internal/100%acme.domain.com from fs1 doesn't ring the phone. Is
there a difference between 'sofia/internal/100%acme.domain.com' and
'user/[EMAIL PROTECTED]'?
Calls out from either fs1 or fs2 routed to the proxy work fine, its just
calls coming in from the proxy. If the call doesn't go to the switch
the user is registered on, the user's phone doesn't ring. It still goes
to voicemail, etc., so that part works.
Is there a better way to cluster FreeSWITCH than DNS SRV records and a
shared state database?
Also, as a side note to Anthony, Brian, et al, if this is the best way,
I'll be happy to write up a wiki page on how I have this setup with a
lot more detail than this. I was not able to find much in the way of
highly available configurations or cluster configurations, so I put
together this system using information cobbled from the wiki, mailing
list messages and lurking on IRC.
Thanks.
- Marc
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Re: [Freeswitch-users] dptools read command doesn't appear to work properly (how to read dtmf via event socket)
those events have nothing to do with the read app the read app is designed to save the dtmf into a variable for you so you would have to wait for the event telling you the execution was complete of read then you can use the command uuid_getvar to get the dtmfdtmf the reason he said you don't need it is because you could just collect it yourself as you get dtmf events and just play the file. On Wed, Oct 29, 2008 at 12:54 PM, Peter P GMX [EMAIL PROTECTED] wrote: I tried to read dtmfs via event socket and came across this thread. What do you mean by: If you're using event socket you have really no reason to use the read application. Is there another chance to do this? Currently I do the following and this isn't successful: I send the following message to FS: SendMsg 4f286c18-a5de-11dd-bef3-4b1a61d55c50 call-command: execute execute-app-name: read execute-app-arg: 0 10 ivr/8000/ivr-enter_ext.wav dtmfdtmf 1 #,* event-lock:true I type 4 digits on the phone 1 2 3 4 plus # I receive 5 separate DTMF events DTMF-Digit: 1 DTMF-Digit: 2 DTMF-Digit: 3 DTMF-Digit: 4 DTMF-Digit: %23 The final event however doesn't deliver the parameter dtmfdtmf as defined above. So I cannot receive the dtmf digits this way. I grepped the network traffic to be sure I haven't missed anything. Any idea? Best regards Peter Brian West schrieb: If you're using event socket you have really no reason to use the read application. The way you invoked it would collect 0 to 4 digits into the variable 'digits'. /b On Sep 12, 2008, at 4:27 PM, Luke Graybill wrote: I am having a hard time getting this command to work properly. Here is the c output from my test session, where I dial in to ext 500 with my sip client (Ekiga): listening on localhost:8084 IPv4Address(TCP, '127.0.0.1 http://127.0.0.1', 36685): Connected 'connect' Caller: Killarny - 'myevents' result was: Events Enabled 'sendmsg\ncall-command: execute\nexecute-app-name: answer' OK 'sendmsg\ncall-command: execute\nexecute-app-name: read\nexecute-app-arg: 0 4 conference/8000/conf-pin.wav digits 1 #' OK DTMF: 1 DTMF: 2 DTMF: 3 DTMF: 4 DTMF: 5 DTMF: 6 DTMF: 7 DTMF: 8 DTMF: 9 DTMF: 0 DTMF: * DTMF: # IPv4Address(TCP, '127.0.0.1 http://127.0.0.1', 36685): Connection was closed cleanly. I couldn't figure out how to get read to function without passing a wav file (I plan to use TTS for the voice prompts, and I don't need a wav file to play here) so I just plugged in a random wav that comes with freeswitch for testing. As you can see, freeswitch reports that the command executed properly, illustrated by the OK (which was a reply-text: +OK) but the read application doesn't seem to be functioning beyond playing the given wav file (and truncating the first half of it as well). Pressing digits in the sip client results in DTMF events being sent to the socket. Note that the DTMF events stop at # because I manually shut down the connection, not because of anything that freeswitch does. Am I doing something wrong here? I feel like I'm following the available documentation given at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read but I am having no luck. Is the read application basically a dead command, and should I be instead just using DTMF event output? Any help that can be provided would be appreciated, thanks. Killarny ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___
Re: [Freeswitch-users] What happend to variable_* in socket_outbound?
Woof! On Wed, 29 Oct 2008 14:18:53 -0400, Michael Jerris [EMAIL PROTECTED] wrote: They should already be on the initial events. Take a look at the raw output, you probably were taking them out of a later event. Nope. Initial event. No variable_* are reported. Using netcat: Connection from 47.16.90.233 port 8084 [tcp/*] accepted connect Channel-Username: 207 Channel-Dialplan: XML Channel-Caller-ID-Name: 207 Channel-Caller-ID-Number: 207 Channel-Network-Addr: 47.16.90.233 Channel-Destination-Number: IVR Channel-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Channel-Source: mod_sofia Channel-Context: default Channel-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Channel-Profile-Index: 1 Channel-Profile-Created-Time: 1225303798001316 Channel-Channel-Created-Time: 1225303798001316 Channel-Channel-Answered-Time: 0 Channel-Channel-Progress-Time: 1225303798010688 Channel-Channel-Progress-Media-Time: 1225303798010688 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Call-Direction: inbound Answer-State: early Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 207 Caller-Dialplan: XML Caller-Caller-ID-Name: 207 Caller-Caller-ID-Number: 207 Caller-Network-Addr: 47.16.90.233 Caller-Destination-Number: IVR Caller-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1225303798001316 Caller-Channel-Created-Time: 1225303798001316 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 1225303798010688 Caller-Channel-Progress-Media-Time: 1225303798010688 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full --Woof! --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] What happend to variable_* in socket_outbound?
yah like i said, update and try again ;) On Wed, Oct 29, 2008 at 1:24 PM, Andy Spitzer [EMAIL PROTECTED] wrote: Woof! On Wed, 29 Oct 2008 14:18:53 -0400, Michael Jerris [EMAIL PROTECTED] wrote: They should already be on the initial events. Take a look at the raw output, you probably were taking them out of a later event. Nope. Initial event. No variable_* are reported. Using netcat: Connection from 47.16.90.233 port 8084 [tcp/*] accepted connect Channel-Username: 207 Channel-Dialplan: XML Channel-Caller-ID-Name: 207 Channel-Caller-ID-Number: 207 Channel-Network-Addr: 47.16.90.233 Channel-Destination-Number: IVR Channel-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Channel-Source: mod_sofia Channel-Context: default Channel-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Channel-Profile-Index: 1 Channel-Profile-Created-Time: 1225303798001316 Channel-Channel-Created-Time: 1225303798001316 Channel-Channel-Answered-Time: 0 Channel-Channel-Progress-Time: 1225303798010688 Channel-Channel-Progress-Media-Time: 1225303798010688 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Call-Direction: inbound Answer-State: early Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 207 Caller-Dialplan: XML Caller-Caller-ID-Name: 207 Caller-Caller-ID-Number: 207 Caller-Network-Addr: 47.16.90.233 Caller-Destination-Number: IVR Caller-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1225303798001316 Caller-Channel-Created-Time: 1225303798001316 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 1225303798010688 Caller-Channel-Progress-Media-Time: 1225303798010688 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full --Woof! --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] What happend to variable_* in socket_outbound?
Woof! On Wed, 29 Oct 2008 14:31:36 -0400, Anthony Minessale [EMAIL PROTECTED] wrote: update and try again ;) Anthony, you are just too fast! Mike was saying one thing, and you checked in the change it while I was replying to him! Updating now. --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Authorizing Anonynous Devices
Ignorance is so Terrible (and of course shameful)! I just create an extension
file in
conf/dialplan/public.
The content of the file is following in case someone cares:
?xml version=1.0 encoding=utf-8?
include
extension name=klaus-extension
condition field=caller_id_number expression=^klaus$
action application=socket data=192.168.50.67:1
full /
/condition
/extension
/include
And to have the calls effectively being matched by this extension i have to
call on port 5080. Thanks Antony for illuminating me.
Klaus.
Original-Nachricht
Datum: Wed, 29 Oct 2008 12:42:45 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Authorizing Anonynous Devices
oh that's good news!
sorry, i assumed you must have had a reason to do it that way.
just create another profile that uses a another port and a public context
with very little extensions in it and no auth options.
then in that public context you can make extens to transfer the call into
your real dialplan when they are appropriate.
the defaults out of the box have something like this in place already
iirc.
On Wed, Oct 29, 2008 at 8:38 AM, Klaus Teller [EMAIL PROTECTED]
wrote:
Sorry if i gave the impression i'm tried to avoid something. There is
nothing i'm trying to avoid, i'm just ignorant.
So how can i translate your recommendation into practice? What
parameters
do i need to set/change?
Thanks,
Klaus.
Original-Nachricht
Datum: Wed, 29 Oct 2008 08:13:58 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Authorizing Anonynous Devices
all this to avoid just making another profile on a different port that
has
inbound calls sandboxed into a special public context?
if you add the port to your srv records nobody would even know.
On Tue, Oct 28, 2008 at 3:02 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi Folks,
I need some additional help with this issue. I already had some from
Brian
i'm but still not able to move forward.
I want a non-registered device to be able to call extension 56900 in
my
Freeswitch in such a way that i can manage the call using the socket
interface. I believe the issue i'm having is that of permissions.
Based on what was suggested sofar, here is what i did.
1) Added following extension in dialplan/default.xml
extension name=check_auth continue=true
condition field=${sip_authorized} expression=^true$
break=never
anti-action application=respond data=407/
/condition
/extension
2) Set auth-calls to false in internal.xml. That is
param name=auth-calls value=$${internal_auth_calls}/
was replaced with:
param name=auth-calls value=false/
3) Changed acl.com.xml by replacing
list name=domains default=DENY
node type=allow domain=$${domain}/
/list
with
list name=domains default=allow
node type=allow domain=$${domain}/
/list
Now here is the result i get after these changes:
a) Anonymous non-registered device can call registered soft phone
at
extension 1003
b) Anonymous non-registered device cannot call 56900 that needs to
be
managed via socket interface (error message 480). Also 9000 cannot
be
called.
c) Registered soft phone (extension 1003) cannot call 56900
d) Registered soft phone (ext 1003) can call registered soft phone
(ext
1000).
If i perform only step 1 and 3 (i.e. auth-calls not set to false),
a)
become impossible, b) remains wrong, c) is now possible (i.e. socket
interface being notified about call at 56900), while d) remains.
valid.
Disabling any of 1) or 3) would result into calls by non-registered
device
being rejected.
Any idea what else can be tried?
Thanks,
Klaus.
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[Freeswitch-users] Using Python to open a channel to outgoing leg and play a prompt
Hi guys, I what I want to do is based on what one of the register extension dials, contact another registered user but before bridging both legs i want to play a file into the egress side, if possible using python Example: call Flow: SIP Phone A -- FreeSwitch -- SIP Phone B. sequence: a) SIP PHONE A dials 777 b) a dialplan is executed based on the user-context, as a condition is matched, then we starts a script ( python is the one i chose ) c) The scripts starts to run and it need to contact the SIP Phone B. when the channel to B is open, I need to play a prompt test.wav to SIP PHONE B, when the prompt is over i do the bridging between SIP PHONE A and SIP PHONE B. If the SIP PHONE B hungs up during the prompt, both channels have to end. I tried with something like: newsession = Session(sofia/internal/[EMAIL PROTECTED]) if ( newsession.ready() ): newsession.execute(playback,test1.wav) but it doesn't work... Do you have some ideas for this... or it is impossible to achieve... -- View this message in context: http://www.nabble.com/Using-Python-to-open-a-channel-to-outgoing-leg-and-play-a-prompt-tp20233448p20233448.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). Michael Jerris wrote: Unsure at this time. There has been some work on mod_cdr_odbc. We generally advise against direct to db cdr methods without a very robust backup method for when the db is down. On Oct 29, 2008, at 9:57 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). Michael Jerris wrote: Unsure at this time. There has been some work on mod_cdr_odbc. We generally advise against direct to db cdr methods without a very robust backup method for when the db is down. On Oct 29, 2008, at 9:57 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
In regards to auto log rotation - YES YES ANTHM just completed that item for me, where by you can set the time in minutes i believe it was. I have not tested it yet, hope to this week. Shawn Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -MC ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Thanks again for your prompt replies, Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -MC ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
That's very good news. :) Shawn Lewis wrote: In regards to auto log rotation - YES YES ANTHM just completed that item for me, where by you can set the time in minutes i believe it was. I have not tested it yet, hope to this week. Shawn Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -MC ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Thanks again for your prompt replies, At that level of activity then I would assume you'd want a more robust solution which obviously would involve a server handling the CDRs separately. That's where XML is a real winner: it can POST CDRs to a web server and the webserver can handle all the pre-processing and db fun stuff. And if the connection to the webserver failed, the CDRs would be put on disk so that they aren't lost forever. Also, the webserver could cache the CDRs to its disk (or whatever storage) if the db itself went down but the webserver stayed up. Just a thought, anyway. It may be extra layers but it's also extra control. -MC Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -MC ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Good point. I have got this kind of behavior (cdrs push model) in my current system (using radius servers). The only drawback of this method is that if you want to be absolutely sure that all the cdrs were handled by the web server (or radius server) you have to check at certain intervals every cdr one by one (and handle those left unhandled for various reasons (network, excessive web server load etc.)) But for my next project I am somewhat forced to use a cdrs-pull method where a process will pull cdrs from the server at its own pace making this extra check unnecessary. I will wait for an automatic log rotation as Shawn Lewis wrote. I think that will do the job. Michael Collins wrote: Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Thanks again for your prompt replies, At that level of activity then I would assume you'd want a more robust solution which obviously would involve a server handling the CDRs separately. That's where XML is a real winner: it can POST CDRs to a web server and the webserver can handle all the pre-processing and db fun stuff. And if the connection to the webserver failed, the CDRs would be put on disk so that they aren't lost forever. Also, the webserver could cache the CDRs to its disk (or whatever storage) if the db itself went down but the webserver stayed up. Just a thought, anyway. It may be extra layers but it's also extra control. -MC Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -MC ___ Freeswitch-users mailing list Freeswitch-users@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
I assume the problem you asked about it happening because the client is disregarding the INVITE from a server with an IP address it was not registered to. If you try to capture the packets going out of your FS (or packets coming in your phone client), I bet you'll see the INVITE request, but no activity thereafter.
I believe that when considering High-Availability for FreeSWITCH, these issues need to be addressed:
1. A shared/floating IP clustering solution such as a load-balancer will only work if the SOFIA hash table is shared between all servers. I donât know if FreeSWITCH entire state is being held in the database or whether some elements are being held in memory.
2. FreeSWITCH needs to have shared-bus architecture to allow for a fully clustered solution. Currently, I donât think that two parked channels on different cluster nodes can be bridged in the current architecture because thereâs no inter-cluster media switching protocol that I know of.
3. A Meshed server approach where different clients are registered to different nodes (like the Cisco Call Manager architecture) seems to be the only immediate option but it is problematic as it requires the client to be configured with a list of redundant servers and most clients donât have that functionality.
4. I would strongly recommend reconsidering the use of any DNS feature (such as SRV records) when deploying a telephony infrastructure. Of course, it all depends what this deployment is for. DNS is commonly used in the ITSP space due to the vast audience but enterprises (all sized) are recommended to refrain adding DNS to the list of point-of-failures in their telephony architectures.
5. I believe the FreeSWITCH conferencing module needs to be adapted to support clustering in order to scale over more than one server. This is due to the same share-bus issue mentioned earlier.
6. In a meshed servers architecture you will need to implement a mechanism that will identify which node in the cluster âownsâ B-Leg, bridge the call to that node and in that node bridge the call again to B-Leg. When you find a way to implement it (I believe FreeSWITCH to have the tools to enable you to do it now), it would solve your current issue.
7. I still have doubts about using carrierroute module opposed to the DISPATCHER module for inbound traffic, mainly because of the registration issue, but I donât have sufficient experience to determine that.
Anyway, itâs very interesting and I definitely like to know how youâre going with it.
On Thu Oct 30 2:04 , "Anthony Minessale" <[EMAIL PROTECTED]>sent:
This all seems right and would make a great wiki page.
What you have described *should* work.
when a phone registers try doing
sofia_contact [EMAIL PROTECTED]
from the cli on each box and see what you get.
you can also use this function in the dialpan
${sofia_contact([EMAIL PROTECTED])}
check that they are both using the same domain name as the profile name
or at least have an alais for it etc.
if it's a bug i can fix it pretty fast as that is the intended behaviour
perhaps you can join irc and get us in the box(s) to have a look at it as we
do not have that situation labbed up anywhere.
On Tue, Oct 28, 2008 at 6:41 PM, Marc Lewis [EMAIL PROTECTED] wrote:
I am in the process of making my FreeSWITCH installation highly
available and I'm running into a couple of snags that was hoping that
someone may have some insight on.
First, the setup as it is now.
There are two installations of FS on two different servers, lets call
them fs1 and fs2. They each pull their configurations, dialplan,
directory and post CDR's all using mod_curl from a central web server.
That part works great.
Calls into and out of FS go through an OpenSER proxy set up using
carrierroute. That part also works great for outbound calls to the
PSTN. Inbound calls also come in through this OpenSER proxy and get
routed to the primary switch fs1. That also works perfectly as long as
its going to fs1.
fs1 and fs2 are both setup to use an ODBC connection to store
registrations. This is pointed to a MySQL database made highly
available using the RedHat Cluster Suite on a shared fibre channel
partition. fs1 and fs2 both share the same database. Voicemail storage
on fs1 is directly mounted on a GFS2 partition, fs2 is mounting the
shared storage from a different server via NFS for no single point of
failure.
For the phones, I have them setup to use SRV records and have fs1 at
priority 10 and fs2 at priority 20 for acme.domain.com. I've tested
this and phones register to the correct server and the sip_registration
table shows either fs1 or fs2 as the hostname as I would expect.
Here is the problem. If user [EMAIL PROTECTED] registers on fs2 and a
call comes in from the OpenSER proxy to fs1, bridging the call to
/sofia/internal/100%acme.domain.com from fs1 doesn't ring the phone. Is
there a difference between 'sofia/internal/100%acme.domain.com' and
'user/[EMAIL PROTECTED]'?
Calls out
Re: [Freeswitch-users] Clustering FreeSWITCH
I'll have to 100% disagree with this statement. NAPTR and SRV are how it should always be done. Toss in some GEO dns and you have many of the problems solved. SRV records should never be optional they should be required to function properly. The NATPR records order preference of records which works in many hard and soft phones. Example which this works: 92:host -t NAPTR bkw.org bkw.org has NAPTR record 10 10 s SIPS+D2T _sips._tcp.bkw.org. bkw.org has NAPTR record 20 20 s SIP+D2S _sip._sctp.bkw.org. bkw.org has NAPTR record 30 30 s SIP+D2T _sip._tcp.bkw.org. bkw.org has NAPTR record 40 40 s SIP+D2U _sip._udp.bkw.org. 93:host -t SRV _sips._tcp.bkw.org _sips._tcp.bkw.org has SRV record 10 0 5061 sip.bkw.org. 94:host -t SRV _sip._sctp.bkw.org. _sip._sctp.bkw.org has SRV record 10 0 5060 sip.bkw.org. 95:host -t SRV _sip._tcp.bkw.org. _sip._tcp.bkw.org has SRV record 10 0 5060 sip.bkw.org. 96:host -t SRV _sip._udp.bkw.org. _sip._udp.bkw.org has SRV record 10 0 5060 sip.bkw.org. With these records in place my Eyebeam will register to my FreeSWITCH instance via TLS since it was listed as the highest preference. The same goes for my Snom phones on my desk. They see the NAPTR's, SRV's and do exactly what I told them to do via DNS. The internet wouldn't exist today without DNS and if your DNS is that fragile you need to figure out why because without it we would be in for a world of hurt Not sure about you but I don't wanna remember what 4 billion IP's go to. Bottom line is NO SRV NO NAPTR you're doing it wrong in my opinion because as a SIP UA you have to look them all up anyway since its NOT optional as per the spec. /b On Oct 29, 2008, at 6:54 PM, Yuval Hertzog wrote: 4. I would strongly recommend reconsidering the use of any DNS feature (such as SRV records) when deploying a telephony infrastructure. Of course, it all depends what this deployment is for. DNS is commonly used in the ITSP space due to the vast audience but enterprises (all sized) are recommended to refrain adding DNS to the list of point-of-failures in their telephony architectures. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
I actually spent a big chunk of today doing various tests. Freeswitch is doing all the right things in this scenario. The problem actually turns out to be my router/firewall that I'm testing behind. When the phones register, it only opens up the port back from the IP address of the server it registers to, so when the secondary server tries to send the invite, the router blocks it. In case anyone is curious, the router/firewall is a Linux box running Arno's Firewall. I'll be doing more tests with different routers to see which ones work and which ones don't. I'll post my results on the wiki page that I'll be creating that covers the setup I've got. For your other points, I'll take them (at least a few of them) one by one. 1. I'm doing this already to an extent. My fs1 box is using a floating IP address and is being monitored using Redhat's cluster suite. If that box goes down, the IP's migrate to a backup machine that contains identical copies of the configurations and access to the shared storage. While not a load balancer, this keeps the primary switch up (except for the wedges that I've been experiencing that I talk about in another thread). The failover switch, my fs2 box, is running on in a Xen guest machine on another server. 2. Freeswitch can't do what you describe. I believe that it does have the architecture for it, though, and it will just be a SMOP(tm) (Simple Matter Of Programming). Once Freeswitch matures a bit more I expect we'll be seeing all sorts of enterprise solutions for it. 3. True. Unless you control everything end to end like Cisco's Call Manager, you have to deal with what's out there, so you work up solutions like the one I've described. 4. Brian followed up on this point, and he said it better than I could. 5. I do agree that conferencing needs to be a bit more robust in a clustered environment. However, there is already a lot of that can be done to make Freeswitch scale by having multiple boxes and putting different conferences on different servers. Using xml_curl, you can write a back-end application that easily routes conferences to multiple different boxes to allow some form of load balancing. 6. I'm not nearly as worried about current calls dropping in the case of a failure as I am about new calls being routed and phones being registered. It would be nice in the case of a failure to not have calls drop, but not a requirement for me. 7. Carrierroute works extremely well for me in my environment. It allows me to have great control with least cost routing as well as have automatic redundant gateways both in and out. It also supports the shared database model for building in my own redundancies. The only thing that I don't like about it is that I can't selectively handle the media path. With my CR setup it doesn't touch any media at all. That has caused me some issues with one or two of my carriers, but nothing that was insurmountable. The ones I've had problems with expect you to be running a b2bua and have media come from the same IP as the SIP messages. For that reason alone I may end up replacing OpenSER with Freeswitch at some point in the future and selectively bypass media, but only if I can get a configuration as efficient as my CR setup. If not, I'll just add a second Freeswitch gateway that talks only to those certain providers. Not ideal, but it works. I will be starting a wiki page about everything I've setup within the next couple days. - Marc Yuval Hertzog wrote: I assume the problem you asked about it happening because the client is disregarding the INVITE from a server with an IP address it was not registered to. If you try to capture the packets going out of your FS (or packets coming in your phone client), I bet you'll see the INVITE request, but no activity thereafter. I believe that when considering High-Availability for FreeSWITCH, these issues need to be addressed: 1. A shared/floating IP clustering solution such as a load-balancer will only work if the SOFIA hash table is shared between all servers. I donâEUR^(TM)t know if FreeSWITCH entire state is being held in the database or whether some elements are being held in memory. 2. FreeSWITCH needs to have shared-bus architecture to allow for a fully clustered solution. Currently, I donâEUR^(TM)t think that two parked channels on different cluster nodes can be bridged in the current architecture because thereâEUR^(TM)s no inter-cluster media switching protocol that I know of. 3. A Meshed server approach where different clients are registered to different nodes (like the Cisco Call Manager architecture) seems to be the only immediate option but it is problematic as it requires the client to be configured with a list of redundant servers and most clients donâEUR^(TM)t have that functionality. 4. I would strongly recommend reconsidering the use of any DNS feature (such as SRV records) when
Re: [Freeswitch-users] Clustering FreeSWITCH
I had this same prblem, so the solution was to re-route the traffic with the info in the sip_registration database via the server the nat¹d client had registered to... This solves the NAT issue... Of couse if we didn¹t have to deal with NAT things would be much easier heh From: Marc Lewis [EMAIL PROTECTED] Organization: Avvatel Corporation Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 19:10:32 -0700 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Clustering FreeSWITCH I actually spent a big chunk of today doing various tests. Freeswitch is doing all the right things in this scenario. The problem actually turns out to be my router/firewall that I'm testing behind. When the phones register, it only opens up the port back from the IP address of the server it registers to, so when the secondary server tries to send the invite, the router blocks it. In case anyone is curious, the router/firewall is a Linux box running Arno's Firewall. I'll be doing more tests with different routers to see which ones work and which ones don't. I'll post my results on the wiki page that I'll be creating that covers the setup I've got. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
what if you turn on that PATH header stuff in openser that we support that lets you pick the reverse proxy path for the calls? On Wed, Oct 29, 2008 at 9:50 PM, Ken Rice [EMAIL PROTECTED] wrote: I had this same prblem, so the solution was to re-route the traffic with the info in the sip_registration database via the server the nat'd client had registered to... This solves the NAT issue... Of couse if we didn't have to deal with NAT things would be much easier heh *From: *Marc Lewis [EMAIL PROTECTED] *Organization: *Avvatel Corporation *Reply-To: *freeswitch-users@lists.freeswitch.org *Date: *Wed, 29 Oct 2008 19:10:32 -0700 *To: *freeswitch-users@lists.freeswitch.org *Subject: *Re: [Freeswitch-users] Clustering FreeSWITCH I actually spent a big chunk of today doing various tests. Freeswitch is doing all the right things in this scenario. The problem actually turns out to be my router/firewall that I'm testing behind. When the phones register, it only opens up the port back from the IP address of the server it registers to, so when the secondary server tries to send the invite, the router blocks it. In case anyone is curious, the router/firewall is a Linux box running Arno's Firewall. I'll be doing more tests with different routers to see which ones work and which ones don't. I'll post my results on the wiki page that I'll be creating that covers the setup I've got. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
That would assume that I was putting openser infront of the boxes... That adds another failure point and yet another set of boxes that end up needing to be redundant From: Anthony Minessale [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 21:55:46 -0500 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Clustering FreeSWITCH what if you turn on that PATH header stuff in openser that we support that lets you pick the reverse proxy path for the calls? On Wed, Oct 29, 2008 at 9:50 PM, Ken Rice [EMAIL PROTECTED] wrote: I had this same prblem, so the solution was to re-route the traffic with the info in the sip_registration database via the server the nat'd client had registered to... This solves the NAT issue... Of couse if we didn't have to deal with NAT things would be much easier heh From: Marc Lewis [EMAIL PROTECTED] Organization: Avvatel Corporation Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 29 Oct 2008 19:10:32 -0700 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Clustering FreeSWITCH I actually spent a big chunk of today doing various tests. Freeswitch is doing all the right things in this scenario. The problem actually turns out to be my router/firewall that I'm testing behind. When the phones register, it only opens up the port back from the IP address of the server it registers to, so when the secondary server tries to send the invite, the router blocks it. In case anyone is curious, the router/firewall is a Linux box running Arno's Firewall. I'll be doing more tests with different routers to see which ones work and which ones don't. I'll post my results on the wiki page that I'll be creating that covers the setup I've got. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
yah i guess, but seems like a proxy as the first thing your traffic hits is not the worst thing though then you can redir the traffic as needed. On Wed, Oct 29, 2008 at 10:03 PM, Ken Rice [EMAIL PROTECTED] wrote: That would assume that I was putting openser infront of the boxes... That adds another failure point and yet another set of boxes that end up needing to be redundant -- *From: *Anthony Minessale [EMAIL PROTECTED] *Reply-To: *freeswitch-users@lists.freeswitch.org *Date: *Wed, 29 Oct 2008 21:55:46 -0500 *To: *freeswitch-users@lists.freeswitch.org *Subject: *Re: [Freeswitch-users] Clustering FreeSWITCH what if you turn on that PATH header stuff in openser that we support that lets you pick the reverse proxy path for the calls? On Wed, Oct 29, 2008 at 9:50 PM, Ken Rice [EMAIL PROTECTED] wrote: I had this same prblem, so the solution was to re-route the traffic with the info in the sip_registration database via the server the nat'd client had registered to... This solves the NAT issue... Of couse if we didn't have to deal with NAT things would be much easier heh *From: *Marc Lewis [EMAIL PROTECTED] *Organization: *Avvatel Corporation *Reply-To: *freeswitch-users@lists.freeswitch.org *Date: *Wed, 29 Oct 2008 19:10:32 -0700 *To: *freeswitch-users@lists.freeswitch.org *Subject: *Re: [Freeswitch-users] Clustering FreeSWITCH I actually spent a big chunk of today doing various tests. Freeswitch is doing all the right things in this scenario. The problem actually turns out to be my router/firewall that I'm testing behind. When the phones register, it only opens up the port back from the IP address of the server it registers to, so when the secondary server tries to send the invite, the router blocks it. In case anyone is curious, the router/firewall is a Linux box running Arno's Firewall. I'll be doing more tests with different routers to see which ones work and which ones don't. I'll post my results on the wiki page that I'll be creating that covers the setup I've got. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 -- ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clustering FreeSWITCH
Marc,
I'll chime in since I'm currently in the process of building a very
similar environment...
I currently have two FS boxes using xml_curl for configuration,
dialplan, and directory data. All sip session info and voicemail data is
stored in the mysql db which is on two multi-master mysql boxes. The two
mysql boxes are in no way clustered, but the DNS A record round robins
between them so generally the FS servers are load balancing their
traffic between the two.. xml_curl pulls its data from the mysql db as
well, so this way I could theoretically add as many FS boxes as I want,
since they will all go back to the db for directory, configuration,
dialplan, voicemail, sip registration data, etc.
The UAs register directly with the FS boxes using DNS SRV and NAPTR
records. As Brian already pointed out, SRV/NAPTR is the best way to go.
Regarding your point of dealing with UAs sitting behind a NAT
firewall/router and registered to any one of your many FS boxes, unless
you have a single proxy for all your UAs, you need to bridge the call to
the FS server the UA is registered with to get through the UA's firewall.
I'm dealing with this in my dialplan through xml_curl. If a call comes
in for a UA, the xml_curl module looks up in the sip_registrations table
the location(s) of the FS server the user is registered with and if
necessary, bridges the call to the appropriate FS server(s). Those
servers in turn look up the user location, realize the user is
registered locally, and generates a ${sofia_contact(user%domain.com
mailto:[EMAIL PROTECTED])}to bridge the call to the one or many
registrations. With UAs behind NAT/firewall routers, I think this is the
only way to do it unless you want a SIP proxy sitting in front of your
FS boxes with a single IP dealing with the UAs.
While this environment isn't completely fault tolerant, I think it's
easily scaled, as you can add more FS boxes with very little
configuration effort since everything goes back to the db.
If you'd like some help putting together the wiki, contact me offline,
I'm more than willing to help.
Now if we could only purchase g.729 licenses for transcoding in FS, that
would solve a huge headache for me :) ...
~Gabe
Marc Lewis wrote:
I actually spent a big chunk of today doing various tests. Freeswitch
is doing all the right things in this scenario. The problem actually
turns out to be my router/firewall that I'm testing behind. When the
phones register, it only opens up the port back from the IP address of
the server it registers to, so when the secondary server tries to send
the invite, the router blocks it. In case anyone is curious, the
router/firewall is a Linux box running Arno's Firewall. I'll be doing
more tests with different routers to see which ones work and which
ones don't. I'll post my results on the wiki page that I'll be
creating that covers the setup I've got.
For your other points, I'll take them (at least a few of them) one by one.
1. I'm doing this already to an extent. My fs1 box is using a
floating IP address and is being monitored using Redhat's cluster
suite. If that box goes down, the IP's migrate to a backup machine
that contains identical copies of the configurations and access to the
shared storage. While not a load balancer, this keeps the primary
switch up (except for the wedges that I've been experiencing that I
talk about in another thread).
The failover switch, my fs2 box, is running on in a Xen guest
machine on another server.
2. Freeswitch can't do what you describe. I believe that it does have
the architecture for it, though, and it will just be a SMOP(tm)
(Simple Matter Of Programming). Once Freeswitch matures a bit more I
expect we'll be seeing all sorts of enterprise solutions for it.
3. True. Unless you control everything end to end like Cisco's Call
Manager, you have to deal with what's out there, so you work up
solutions like the one I've described.
4. Brian followed up on this point, and he said it better than I could.
5. I do agree that conferencing needs to be a bit more robust in a
clustered environment. However, there is already a lot of that can be
done to make Freeswitch scale by having multiple boxes and putting
different conferences on different servers. Using xml_curl, you can
write a back-end application that easily routes conferences to
multiple different boxes to allow some form of load balancing.
6. I'm not nearly as worried about current calls dropping in the case
of a failure as I am about new calls being routed and phones being
registered. It would be nice in the case of a failure to not have
calls drop, but not a requirement for me.
7. Carrierroute works extremely well for me in my environment. It
allows me to have great control with least cost routing as well as
have automatic redundant gateways both in and out. It also supports
the shared database model for building in my own redundancies. The
only thing that I don't like about it
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
[EMAIL PROTECTED] wrote: Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Not that you'd notice. We run XML CDR to database scripting on each box that we use for switching, and it's a pretty trivial task compared with switching all that media. Doing it this way is:- (a) distributed - one process per box scales nicely; (b) robust - script down, DB down, no problem: files just queue up; (c) simple - the script logic is trivial: - while 1 - for each file in the XML CDR directory - open it - parse it (XML::Simple for us) - insert it in to the DB - delete it - sleep for a couple of seconds Two error cases: can't parse or can't find data which should be there: move the file in to another directory to be examined by real eyes; DB insert fails: break out of inner loop and it'll be retried after a short pause. --Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
