[Freeswitch-users] [CentOS] Modifying init script?
Hello
I'm installing Freeswitch on a CentOS 5.3 test host. I noticed that the
freeswitch.init.redhat contains paths that don't match a stock CentOS
install.
All files are currently installed/owned by root: Should I go ahead, create a
freeswitch user/group, and chown everything under /usr/local/freeswitch/,
or are there issues if some files aren't owned by root?
Here's the original file, and mine:
= ORIG /usr/src/freeswitch/build/freeswitch.init.redhat =
PROG_NAME=freeswitch
PID_FILE=${PID_FILE-/opt/freeswitch/log/freeswitch.pid}
FS_USER=${FS_USER-freeswitch}
FS_FILE=${FS_FILE-/opt/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/opt/freeswitch}
LOCK_FILE=/var/lock/subsys/freeswitch
FREESWITCH_ARGS=-nc
RETVAL=0
= My /etc/init.d/freeswitch =
PROG_NAME=freeswitch
PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid}
#All files installed/owned by root: Should I go ahead and create a
freeswitch user anyway?
FS_USER=${FS_USER-freeswitch}
FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/usr/local/freeswitch}
LOCK_FILE=/var/lock/subsys/freeswitch
FREESWITCH_ARGS=-nc
RETVAL=0
==
Thank you.
--
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[Freeswitch-users] How to use variables in console/event socket
Hello,
I don't know how to use variables in console. I try to call local user with
following commands.
Command: originate sofia/internal/1001%${domain}
I get: Cannot locate registered user 1...@%{domain}
Command: originate sofia/internal/1...@${domain}
I get: Cannot locate registered user 1001@@{domain}
Command: originate sofia/internal/1001%\${domain}
I get: Cannot locate registered user 1...@\%{domain}
Substitution of $ sign is not understood for me. Is it possible to use
variables in that channels?
Variable is defined:
freeswi...@vertux global_getvar domain
API CALL [global_getvar(domain)] output:
192.168.77.248
Szymon O.
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[Freeswitch-users] ACL not working
Hey guys,
I'm currently testing FS inside a LAN. FreeSWITCH is running on
192.168.0.101 and my softphone is on 192.168.0.100.
I can register and make calls just fine, but I want to deny everything in
order to learn how the ACL works.
I have this on the internal profile:
param name=apply-nat-acl value=rfc1918/
param name=apply-inbound-acl value=domains/
param name=apply-register-acl value=domains/
And this is how my acl.conf.xml looks, it's all set to deny:
configuration name=acl.conf description=Network Lists
network-lists
list name=dl-candidates default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=rfc1918 default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=lan default=deny
node type=deny cidr=192.168.42.0/24/
node type=deny cidr=192.168.42.42/32/
/list
list name=strict default=deny
node type=deny cidr=208.102.123.124/32/
/list
!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
--
list name=domains default=deny
node type=deny domain=$${domain}/
node type=deny cidr=192.168.0.0/24/
/list
/network-lists
/configuration
But I'm still allowed to register with the 1000 user and make calls, to the
conference extension, etc... I can't understand this, if it's all to deny
and the cidr is set to 192.168.0.0/24 on the domains context, which is
what hte profile uses, shouldn't the registration/call be denied. I have
tried many conbinations but whenever I change something it wont make any
difference.
Please help me.
Thanks,
Diego
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Re: [Freeswitch-users] ACL not working
If I make any changes on the acl.conf.xml, it doesn't take any effect.
Why is that? What am I doing wrong?
Diego
On Tue, Apr 21, 2009 at 5:29 AM, Diego Viola diego.vi...@gmail.com wrote:
More info:
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
!-- param name=accept-blind-reg value=true/ --
!-- param name=accept-blind-auth value=true/ --
So any ideas?
On Tue, Apr 21, 2009 at 5:08 AM, Diego Viola diego.vi...@gmail.comwrote:
Hey guys,
I'm currently testing FS inside a LAN. FreeSWITCH is running on
192.168.0.101 and my softphone is on 192.168.0.100.
I can register and make calls just fine, but I want to deny everything in
order to learn how the ACL works.
I have this on the internal profile:
param name=apply-nat-acl value=rfc1918/
param name=apply-inbound-acl value=domains/
param name=apply-register-acl value=domains/
And this is how my acl.conf.xml looks, it's all set to deny:
configuration name=acl.conf description=Network Lists
network-lists
list name=dl-candidates default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=rfc1918 default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=lan default=deny
node type=deny cidr=192.168.42.0/24/
node type=deny cidr=192.168.42.42/32/
/list
list name=strict default=deny
node type=deny cidr=208.102.123.124/32/
/list
!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
--
list name=domains default=deny
node type=deny domain=$${domain}/
node type=deny cidr=192.168.0.0/24/
/list
/network-lists
/configuration
But I'm still allowed to register with the 1000 user and make calls, to
the conference extension, etc... I can't understand this, if it's all to
deny and the cidr is set to 192.168.0.0/24 on the domains context,
which is what hte profile uses, shouldn't the registration/call be denied. I
have tried many conbinations but whenever I change something it wont make
any difference.
Please help me.
Thanks,
Diego
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Re: [Freeswitch-users] ACL not working
freeswi...@internal acl
false
On Tue, Apr 21, 2009 at 5:08 AM, Diego Viola diego.vi...@gmail.com wrote:
Hey guys,
I'm currently testing FS inside a LAN. FreeSWITCH is running on
192.168.0.101 and my softphone is on 192.168.0.100.
I can register and make calls just fine, but I want to deny everything in
order to learn how the ACL works.
I have this on the internal profile:
param name=apply-nat-acl value=rfc1918/
param name=apply-inbound-acl value=domains/
param name=apply-register-acl value=domains/
And this is how my acl.conf.xml looks, it's all set to deny:
configuration name=acl.conf description=Network Lists
network-lists
list name=dl-candidates default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=rfc1918 default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=lan default=deny
node type=deny cidr=192.168.42.0/24/
node type=deny cidr=192.168.42.42/32/
/list
list name=strict default=deny
node type=deny cidr=208.102.123.124/32/
/list
!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
--
list name=domains default=deny
node type=deny domain=$${domain}/
node type=deny cidr=192.168.0.0/24/
/list
/network-lists
/configuration
But I'm still allowed to register with the 1000 user and make calls, to the
conference extension, etc... I can't understand this, if it's all to deny
and the cidr is set to 192.168.0.0/24 on the domains context, which is
what hte profile uses, shouldn't the registration/call be denied. I have
tried many conbinations but whenever I change something it wont make any
difference.
Please help me.
Thanks,
Diego
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Re: [Freeswitch-users] Best G729 replacement
I don't think it's a Tier 1. They are middlemen between phone companies and other companies that need voip. In my country (Chile), phone companies don't provide voip services, so you have to buy the service from someone else. On Tue, Apr 21, 2009 at 12:45 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: On Mon, Apr 20, 2009 at 3:10 PM, Nicolas Brenner nico...@medularis.com wrote: Hi, I might be in a position (finally) to ask/suggest one of my voip providers to use an alternative codec to G729. I wanted to know what would be the best replacement for it. Thanks again everybody for your time and info. Regards, Nicolas Nicolas, What do you mean by provider? Is this a Tier 1? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 ___ 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] ACL not working
Ok I just remade the config and now it's working as it should, it's not
letting me register.
2009-04-21 07:06:03 [WARNING] sofia_reg.c:1283
sofia_reg_handle_sip_i_register() IP 192.168.0.100 Rejected by acl domains
However, I have this:
param name=apply-inbound-acl value=domains/
And this:
list name=domains default=deny
!-- node type=allow domain=$${domain}/--
node type=deny cidr=192.168.0.100/32/
node type=deny cidr=192.168.0.0/24/
/list
And I can still call the conference (3030) without being registered. Why is
this?
Thanks.
On Tue, Apr 21, 2009 at 6:43 AM, Diego Viola diego.vi...@gmail.com wrote:
freeswi...@internal acl
false
On Tue, Apr 21, 2009 at 5:08 AM, Diego Viola diego.vi...@gmail.comwrote:
Hey guys,
I'm currently testing FS inside a LAN. FreeSWITCH is running on
192.168.0.101 and my softphone is on 192.168.0.100.
I can register and make calls just fine, but I want to deny everything in
order to learn how the ACL works.
I have this on the internal profile:
param name=apply-nat-acl value=rfc1918/
param name=apply-inbound-acl value=domains/
param name=apply-register-acl value=domains/
And this is how my acl.conf.xml looks, it's all set to deny:
configuration name=acl.conf description=Network Lists
network-lists
list name=dl-candidates default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=rfc1918 default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=lan default=deny
node type=deny cidr=192.168.42.0/24/
node type=deny cidr=192.168.42.42/32/
/list
list name=strict default=deny
node type=deny cidr=208.102.123.124/32/
/list
!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
--
list name=domains default=deny
node type=deny domain=$${domain}/
node type=deny cidr=192.168.0.0/24/
/list
/network-lists
/configuration
But I'm still allowed to register with the 1000 user and make calls, to
the conference extension, etc... I can't understand this, if it's all to
deny and the cidr is set to 192.168.0.0/24 on the domains context,
which is what hte profile uses, shouldn't the registration/call be denied. I
have tried many conbinations but whenever I change something it wont make
any difference.
Please help me.
Thanks,
Diego
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Re: [Freeswitch-users] Quick question on how XML files are handled
mercutioviz wrote: Anyway, the bottom line is that any XML file in autoload_configs/ will get loaded into the big XML configuration file that gets stored in memory, even if the corresponding module is not loaded from modules.conf.xml. So all configuration settings found in autoload_configs/*.xml files are loaded in memory, but Freeswitch will only actually use those that belong to modules that are explicitely loaded through the modules.conf.xml file. Makes sense. Thanks guys. -- View this message in context: http://www.nabble.com/Quick-question-on-how-XML-files-are-handled-tp23136269p23153843.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
[Freeswitch-users] Help with mod_managed under Windows
I am playing around with FS (Windows) for one month now. First I tried using FreeSwitch.NET which is a good class library for inbound event socket. Unfortunatley it can't be used for outbound event socket. So I read the wiki back ond forth and also searched the net and found that I should use mod_managed. So I downloaded mod_managed in source from svn and compiled it with C# 2008 Express Edition. After that I got a dll. FreeSwitch.Managed.dll and copied it to the mod dir of FS. The Problem is that I get and error when FS loads mod_managed and I don't know what I should do with that. 2009-04-21 11:26:44 [INFO] mod_managed.cpp:314 mod_managed_load() Loading mod_ma naged (Common Language Infrastructure), Microsoft CLR Version 2009-04-21 11:26:44 [ERR] mod_managed.cpp:333 mod_managed_load() Load did not re turn true. System.Reflection.TargetInvocationException: Ein Aufrufziel hat einen Ausnahmefehler verursacht. --- System.TypeInitializationException: Der Typenin itialisierer f³r FreeSWITCH.Native.freeswitch hat eine Ausnahme verursacht. -- - System.EntryPointNotFoundException: Der Einstiegspunkt CSharp_SWITCH_READ_TE RMINATOR_USED_VARIABLE_get wurde nicht in der DLL mod_managed gefunden. bei FreeSWITCH.Native.freeswitchPINVOKE.SWITCH_READ_TERMINATOR_USED_VARIABLE_ get() bei FreeSWITCH.Native.freeswitch..cctor() --- Ende der internen Ausnahmestapel³berwachung --- bei FreeSWITCH.Native.freeswitch.get_SWITCH_GLOBAL_dirs() bei FreeSWITCH.Loader.Load() --- Ende der internen Ausnahmestapel³berwachung --- bei System.RuntimeMethodHandle._InvokeMethodFast(Object target, Object[] argu ments, SignatureStruct sig, MethodAttributes methodAttributes, RuntimeTypeHandl e typeOwner) bei System.RuntimeMethodHandle.InvokeMethodFast(Object target, Object[] argum ents, Signature sig, MethodAttributes methodAttributes, RuntimeTypeHandle typeOw ner) bei System.Reflection.RuntimeMethodInfo.Invoke(Object obj, BindingFlags invok eAttr, Binder binder, Object[] parameters, CultureInfo culture, Boolean skipVisi bilityChecks) bei System.Reflection.RuntimeMethodInfo.Invoke(Object obj, BindingFlags invok eAttr, Binder binder, Object[] parameters, CultureInfo culture) bei mod_managed_load(switch_loadable_module_interface** module_interface, apr _pool_t* pool) 2009-04-21 11:26:44 [CRIT] switch_loadable_module.c:845 switch_loadable_module_l oad_file() Error Loading module C:\Programme\FreeSWITCH\mod\mod_managed.dll **Module load routine returned an error** Please help me with that. Thanks...Guido ___ 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] javascript: session.recordFile
I'm using session.recordFile in a javascript. How do I check the length of the recorded file? ___ 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] ACL not working
Oh it was because I had auth-calls set to true, now I turned it false and it
works as I expect!
Silly me, thanks everyone anyway =D
Diego
On Tue, Apr 21, 2009 at 7:08 AM, Diego Viola diego.vi...@gmail.com wrote:
Ok I just remade the config and now it's working as it should, it's not
letting me register.
2009-04-21 07:06:03 [WARNING] sofia_reg.c:1283
sofia_reg_handle_sip_i_register() IP 192.168.0.100 Rejected by acl domains
However, I have this:
param name=apply-inbound-acl value=domains/
And this:
list name=domains default=deny
!-- node type=allow domain=$${domain}/--
node type=deny cidr=192.168.0.100/32/
node type=deny cidr=192.168.0.0/24/
/list
And I can still call the conference (3030) without being registered. Why is
this?
Thanks.
On Tue, Apr 21, 2009 at 6:43 AM, Diego Viola diego.vi...@gmail.comwrote:
freeswi...@internal acl
false
On Tue, Apr 21, 2009 at 5:08 AM, Diego Viola diego.vi...@gmail.comwrote:
Hey guys,
I'm currently testing FS inside a LAN. FreeSWITCH is running on
192.168.0.101 and my softphone is on 192.168.0.100.
I can register and make calls just fine, but I want to deny everything in
order to learn how the ACL works.
I have this on the internal profile:
param name=apply-nat-acl value=rfc1918/
param name=apply-inbound-acl value=domains/
param name=apply-register-acl value=domains/
And this is how my acl.conf.xml looks, it's all set to deny:
configuration name=acl.conf description=Network Lists
network-lists
list name=dl-candidates default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=rfc1918 default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=lan default=deny
node type=deny cidr=192.168.42.0/24/
node type=deny cidr=192.168.42.42/32/
/list
list name=strict default=deny
node type=deny cidr=208.102.123.124/32/
/list
!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
--
list name=domains default=deny
node type=deny domain=$${domain}/
node type=deny cidr=192.168.0.0/24/
/list
/network-lists
/configuration
But I'm still allowed to register with the 1000 user and make calls, to
the conference extension, etc... I can't understand this, if it's all to
deny and the cidr is set to 192.168.0.0/24 on the domains context,
which is what hte profile uses, shouldn't the registration/call be denied. I
have tried many conbinations but whenever I change something it wont make
any difference.
Please help me.
Thanks,
Diego
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Re: [Freeswitch-users] ACL not working
More info:
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
!-- param name=accept-blind-reg value=true/ --
!-- param name=accept-blind-auth value=true/ --
So any ideas?
On Tue, Apr 21, 2009 at 5:08 AM, Diego Viola diego.vi...@gmail.com wrote:
Hey guys,
I'm currently testing FS inside a LAN. FreeSWITCH is running on
192.168.0.101 and my softphone is on 192.168.0.100.
I can register and make calls just fine, but I want to deny everything in
order to learn how the ACL works.
I have this on the internal profile:
param name=apply-nat-acl value=rfc1918/
param name=apply-inbound-acl value=domains/
param name=apply-register-acl value=domains/
And this is how my acl.conf.xml looks, it's all set to deny:
configuration name=acl.conf description=Network Lists
network-lists
list name=dl-candidates default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=rfc1918 default=deny
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
list name=lan default=deny
node type=deny cidr=192.168.42.0/24/
node type=deny cidr=192.168.42.42/32/
/list
list name=strict default=deny
node type=deny cidr=208.102.123.124/32/
/list
!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
--
list name=domains default=deny
node type=deny domain=$${domain}/
node type=deny cidr=192.168.0.0/24/
/list
/network-lists
/configuration
But I'm still allowed to register with the 1000 user and make calls, to the
conference extension, etc... I can't understand this, if it's all to deny
and the cidr is set to 192.168.0.0/24 on the domains context, which is
what hte profile uses, shouldn't the registration/call be denied. I have
tried many conbinations but whenever I change something it wont make any
difference.
Please help me.
Thanks,
Diego
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Re: [Freeswitch-users] ACL not working
Do you want to allow these IP ranges? /b On Apr 21, 2009, at 6:08 AM, Diego Viola wrote: node type=deny cidr=192.168.0.100/32/ node type=deny cidr=192.168.0.0/24/ Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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
[Freeswitch-users] getting statistics from core db through esl interface
Hi, I'm making a c plugin for collectd ( http://collectd.org ) to get some basic statistics from FS. Right now it uses ESL to connect, but doing an api show channels and after that parsing the csv to get the amount of channels per profile seems a bit of a detour.. Is it possible to do some sql selects on the core db through ESL to get this info ? thanks regards, Leon ___ 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] How to use variables in console/event socket
Variables aren't expanded on the command line. You'll have to put exactly what you want in. /b On Apr 21, 2009, at 3:48 AM, Szymon Olko wrote: Substitution of $ sign is not understood for me. Is it possible to use variables in that channels? Variable is defined: freeswi...@vertux global_getvar domain API CALL [global_getvar(domain)] output: 192.168.77.248 Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] getting statistics from core db through esl interface
Thanks, I know that's possible, but I want to get the amount of channels per profile (and even seperated inbound/outbound), not a total of all channels on all profiles combined.. regards, Leon On Apr 21, 2009, at 3:41 PM, Mathieu Rene wrote: show channels count show calls count Mathieu On 21-Apr-09, at 9:27 AM, Leon de Rooij wrote: Hi, I'm making a c plugin for collectd ( http://collectd.org ) to get some basic statistics from FS. Right now it uses ESL to connect, but doing an api show channels and after that parsing the csv to get the amount of channels per profile seems a bit of a detour.. Is it possible to do some sql selects on the core db through ESL to get this info ? thanks regards, Leon ___ 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] getting statistics from core db through esl interface
You forgot the as xml versions show channels as xml show calls count as xml show channels count as xml ;) /b On Apr 21, 2009, at 8:41 AM, Mathieu Rene wrote: show channels count show calls count Mathieu Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] which freeswitch sip trunk is gd for testing
Prabhuram Mohan, Give Teliax a try. https://teliax.com/?referral_code=11 Our pay as you go plan would be an excellent choice. It includes 10 channels and you only pay for the usage and the DID. Please use referral code 11 while signing up for service. We also offer wholesale rates for those looking for large scale deployments. If you have questions please contact me. Kind Regards, Geoff Love 303-629-8304 gl...@teliax.com Referral code 11 On Mon, Apr 20, 2009 at 9:55 PM, Prabhuram Mohan mprabhu...@gmail.comwrote: yo! i am working on sip appliation. i want to buy 1 or 2 sip trunks to be added and test a functionality. i thought of buying voicepulse.com but the reviews were not gd.. do you have any suggestion.. i wanna use this trunk for testing only.. ___ 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 -- Geoff Love Sales Engineer gl...@teliax.com 303-629-8304 Referral Code 11 ___ 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] How to tell if 100 Trying received
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of
gateways that can make the call. However, if one of them is down or has
some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it
hasn't been received in a couple of seconds then go on and try the next
gateway.
But I can't find a flag/event/state which corresponds to receipt of this
message. Can anyone tell me where I should be looking? I put a debug print
in sofia_event_callback for every event but there doesn't seem to be one
fired for this condition.
Thanks in advance.
{P^/
John
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[Freeswitch-users] VIA messages?
Hello I'm reading the examples, and found one on how to use a Sipura SPA-2000: http://wiki.freeswitch.org/wiki/Sipura_STUN I was wondering what the VIA messages mean, and whether I should enable those when the unit is used behind a NAT router. I thought Freeswitch would take care of NAT, but it looks like SIP devices have to be configured for this. Anybody knows what VIA messages are for? Thank you. -- View this message in context: http://www.nabble.com/VIA-messages--tp23157781p23157781.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] How to tell if 100 Trying received
That 100 trying is handled deep in the sip stack.
The author of sofia said it would be a big job to bring that up to the even
callback.
Someone may be able to persuade him to allow you to pass a global timeout
waiting for 100
or something but no solution exists atm
On Tue, Apr 21, 2009 at 3:32 AM, John Dalgliesh jo...@defyne.org wrote:
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of
gateways that can make the call. However, if one of them is down or has
some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it
hasn't been received in a couple of seconds then go on and try the next
gateway.
But I can't find a flag/event/state which corresponds to receipt of this
message. Can anyone tell me where I should be looking? I put a debug print
in sofia_event_callback for every event but there doesn't seem to be one
fired for this condition.
Thanks in advance.
{P^/
John
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] which freeswitch sip trunk is gd for testing
Prabhuram, I would recommend Teliax. I've been with them for over 5 years now, originally with Asterisk and now with Freeswitch. I don't know of any better provider. On Tue, 21 Apr 2009 08:25:18 -0600, Geoff Love gl...@teliax.com wrote: Prabhuram Mohan, Give Teliax a try. https://teliax.com/?referral_code=11 Our pay as you go plan would be an excellent choice. It includes 10 channels and you only pay for the usage and the DID. Please use referral code 11 while signing up for service. We also offer wholesale rates for those looking for large scale deployments. If you have questions please contact me. Kind Regards, Geoff Love 303-629-8304 gl...@teliax.com Referral code 11 On Mon, Apr 20, 2009 at 9:55 PM, Prabhuram Mohan mprabhu...@gmail.comwrote: yo! i am working on sip appliation. i want to buy 1 or 2 sip trunks to be added and test a functionality. i thought of buying voicepulse.com but the reviews were not gd.. do you have any suggestion.. i wanna use this trunk for testing only.. ___ 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 -- Kenneth Shaw ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949.650.4600 fax: 949.642.6044 k...@expitrans.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] Configure FS using Flowroute.com
You've called your gateway flowroute vs. mine which is sip.flowroute.com;
you need to amend the dialplan to reference flowroute; then you should be set.
flowroute gateway sip:...@sip.flowroute.com
action application=bridge
data=sofia/gateway/flowroute/${default_provider_username}#$1/
C.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex Harris
Sent: Monday, April 20, 2009 20:22
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
The vars.xml has been modified with the following:
X-PRE-PROCESS cmd=set data=default_provider=sip.flowroute.com/
X-PRE-PROCESS cmd=set data=default_provider_username=***/
X-PRE-PROCESS cmd=set data=default_provider_password=/
X-PRE-PROCESS cmd=set
data=default_provider_from_domain=sip.flowroute.com/
!-- true or false --
X-PRE-PROCESS cmd=set data=default_provider_register=true/
X-PRE-PROCESS cmd=set data=default_provider_contact=5000/
the 01_example.xml has been modified with the code you provided
The Status output is:
Name Type Data
State
===
internal profile sip:mod_so...@**.***.219.221:5060
RUNNING (0)
external profile sip:mod_so...@**.***.219.221:5080
RUNNING (0)
flowroute gateway sip:...@sip.flowroute.com
REGED
**.***.219.221 alias internal
ALIASED
internal-ipv6 profile
sip:mod_so...@[2002:63be:dbdd::63be:dbdd]:5060RUNNING (0)
default alias internal
ALIASED
nat alias external
ALIASED
outbound alias external
ALIASED
==
When I try to dial out this is what I am getting:
freeswi...@s4bs 2009-04-20 20:19:45 [NOTICE] switch_channel.c:592
switch_channel_set_name() New Channel sofia/internal/1...@**.***.219.221
[7bbb17eb-e953-1743-84b5-d4b0ae651332] 2009-04-20 20:19:45 [INFO]
mod_dialplan_xml.c:252 dialplan_hunt() Processing Alex-15184282539 in
context default 2009-04-20 20:19:45 [ERR] mod_sofia.c:2411
sofia_outgoing_channel() Invalid Gateway 2009-04-20 20:19:45 [NOTICE]
mod_sofia.c:2624 sofia_outgoing_channel() Close Cha
nnel N/A [CS_NEW]
2009-04-20 20:19:45 [ERR] switch_ivr_originate.c:1459 switch_ivr_originate()
Can
not create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2009-04-20 20:19:45 [INFO] mod_dptools.c:2036 audio_bridge_function()
Originate
Failed. Cause: INVALID_NUMBER_FORMAT
2009-04-20 20:19:45 [NOTICE] mod_dptools.c:2068 audio_bridge_function()
Hangup s
ofia/internal/1...@**.***.219.221 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1018
switch_core_session_thre
ad() Session 3 (sofia/internal/1...@**.***.219.221) Ended
2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1020
switch_core_session_thre
ad() Close Channel sofia/internal/1...@**.***.219.221 [CS_DONE]
Thank you for the help.
~Alex
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Fowler
Sent: Monday, April 20, 2009 7:09 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
Are you either restarting FS or issuing the reloadxml command (press F6 on
the console) after making these changes?
Did you modify vars.xml per my last note?
Also, check out http://wiki.freeswitch.org/wiki/Getting_Started_Guide - it's
worth investing the time to understand how FS parses the various config
files. Look in /usr/local/freeswitch/log at freeswitch.xml.fsxml - this
file contains the entire free switch configuration since last as parsed by
the application.
Seems you're not hitting the issue I was with FlowRoute - but as the error
indicates you're trying to route a call out from a gateway that does not
exist.
What's the output of sofia status (F5 on the console)? It should show:
sip.flowroute.com gateway sip:...@sip.flowroute.com
REGED
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex
Harris
Sent: Monday, April 20, 2009 17:05
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
I modified the 01_example.com.xml with the code below and I am still getting
Invalid Gateway. I tried creating a file 01_flowroute.com.xml and placed
the
Re: [Freeswitch-users] Configure FS using Flowroute.com
That did it! Thank you so much for all your help.
~Alex
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Fowler
Sent: Tuesday, April 21, 2009 8:54 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
You've called your gateway flowroute vs. mine which is
sip.flowroute.com; you need to amend the dialplan to reference
flowroute; then you should be set.
flowroute gateway
sip:...@sip.flowroute.com
action application=bridge
data=sofia/gateway/flowroute/${default_provider_username}#$1/
C.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alex
Harris
Sent: Monday, April 20, 2009 20:22
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
The vars.xml has been modified with the following:
X-PRE-PROCESS cmd=set data=default_provider=sip.flowroute.com/
X-PRE-PROCESS cmd=set data=default_provider_username=***/
X-PRE-PROCESS cmd=set data=default_provider_password=/
X-PRE-PROCESS cmd=set
data=default_provider_from_domain=sip.flowroute.com/
!-- true or false --
X-PRE-PROCESS cmd=set data=default_provider_register=true/
X-PRE-PROCESS cmd=set data=default_provider_contact=5000/
the 01_example.xml has been modified with the code you provided
The Status output is:
Name Type Data
State
===
internal profile sip:mod_so...@**.***.219.221:5060
RUNNING (0)
external profile sip:mod_so...@**.***.219.221:5080
RUNNING (0)
flowroute gateway sip:...@sip.flowroute.com
REGED
**.***.219.221 alias internal
ALIASED
internal-ipv6 profile
sip:mod_so...@[2002:63be:dbdd::63be:dbdd]:5060RUNNING (0)
default alias internal
ALIASED
nat alias external
ALIASED
outbound alias external
ALIASED
==
When I try to dial out this is what I am getting:
freeswi...@s4bs 2009-04-20 20:19:45 [NOTICE] switch_channel.c:592
switch_channel_set_name() New Channel sofia/internal/1...@**.***.219.221
[7bbb17eb-e953-1743-84b5-d4b0ae651332] 2009-04-20 20:19:45 [INFO]
mod_dialplan_xml.c:252 dialplan_hunt() Processing Alex-15184282539 in
context default 2009-04-20 20:19:45 [ERR] mod_sofia.c:2411
sofia_outgoing_channel() Invalid Gateway 2009-04-20 20:19:45 [NOTICE]
mod_sofia.c:2624 sofia_outgoing_channel() Close Cha
nnel N/A [CS_NEW]
2009-04-20 20:19:45 [ERR] switch_ivr_originate.c:1459 switch_ivr_originate()
Can
not create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2009-04-20 20:19:45 [INFO] mod_dptools.c:2036 audio_bridge_function()
Originate
Failed. Cause: INVALID_NUMBER_FORMAT
2009-04-20 20:19:45 [NOTICE] mod_dptools.c:2068 audio_bridge_function()
Hangup s
ofia/internal/1...@**.***.219.221 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1018
switch_core_session_thre
ad() Session 3 (sofia/internal/1...@**.***.219.221) Ended
2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1020
switch_core_session_thre
ad() Close Channel sofia/internal/1...@**.***.219.221 [CS_DONE]
Thank you for the help.
~Alex
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Fowler
Sent: Monday, April 20, 2009 7:09 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com
Are you either restarting FS or issuing the reloadxml command (press F6 on
the console) after making these changes?
Did you modify vars.xml per my last note?
Also, check out http://wiki.freeswitch.org/wiki/Getting_Started_Guide - it's
worth investing the time to understand how FS parses the various config
files. Look in /usr/local/freeswitch/log at freeswitch.xml.fsxml - this
file contains the entire free switch configuration since last as parsed by
the application.
Seems you're not hitting the issue I was with FlowRoute - but as the error
indicates you're trying to route a call out from a gateway that does not
exist.
What's the output of sofia status (F5 on the console)? It should show:
sip.flowroute.com gateway sip:...@sip.flowroute.com
REGED
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Re: [Freeswitch-users] javascript: session.recordFile
On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: I'm using session.recordFile in a javascript. How do I check the length of the recorded file? You mean once the file is recorded you want to know how many minutes/seconds long it is? -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] VoiceXML
Hi everyone, Did someone ever try and implement a VXML interface on freeswitch? Or do you think it's a good idea? Or not? Since it is an actual standard, I guess there might be a market for application service providers. Remko ___ 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] fifo taking 5 seconds to bridge calls
There was never really a problem it was just a minor behavioral change. If you have it in production and need such careful support, you should send an email to consult...@freeswitch.org and sign up for a commercial support contract. On Tue, Apr 21, 2009 at 12:37 AM, Matthew Fong mattdf...@gmail.com wrote: Hi Anthony Brian, I have not yet run r13094 in my production environment with live agents, so I cannot give you feedback (but hopefully I'll get a chance to put it into a live system in a few days) but I re-reviewed the logs I had and I'm not convinced the issue I was having of a delayed bridge was related to the default fifo_consumer_wrapup_time. The reason is: 1) it's not a consistent 5 second delay in bridging...sometimes it's a 2 second delay, sometimes it's as high as a 38 second delay (I can provide logs if needed) 2) In my setup, each consumer (channel that executes fifo out) is always a fresh/new channel. My consumers do not get recycled, instead they get hungup at the end of the call (while my fifo ins get transferred to another extension, which puts them back into fifo in) Are these problems still consistent with the issues that were fixed in r13094? I'm a little hesitant to put the system back in a live environment since the fix and diagnosis aren't 100% compatible. As always tho, thanks for the really quick fix and reply. Awesome telephone framework. --matt On Tue, Apr 21, 2009 at 9:53 AM, Matthew Fong mattdf...@gmail.com wrote: Thanks I'll check it out. One more quick but related question. Is there ever an instance when the audio is BRIDGED before the BRIDGE event is fired. Could this fifo issue have bridged audio immediately, but somehow withheld the bridge event from being fired for 5 seconds? A few of my callers were reporting they could hear the Contact, but the BRIDGE event (and my subsequent programming to popup the contact information on screen) was being delayed 5 seconds. thanks ! --matt On Tue, Apr 21, 2009 at 9:23 AM, Brian West br...@freeswitch.org wrote: Update rev. 13094 makes it not do wrap up on nowait. /b On Apr 20, 2009, at 8:02 AM, Anthony Minessale wrote: it's probably the designed wrapup time for agents. fifo_consumer_wrapup_time var controls this wait time in milliseconds and the default is 5 sec. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] javascript: session.recordFile
Yes, to be able to determine if it's too short or not. On Tue, Apr 21, 2009 at 7:01 PM, Michael Collins m...@freeswitch.org wrote: On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: I'm using session.recordFile in a javascript. How do I check the length of the recorded file? You mean once the file is recorded you want to know how many minutes/seconds long it is? -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] VoiceXML
To the best of my knowledge no one has created an interface for this. I'm sure one will be created if and when the market demand gets high enough. -MC On Tue, Apr 21, 2009 at 10:57 AM, Remko Kloosterman r.klooster...@mtel.nlwrote: Hi everyone, Did someone ever try and implement a VXML interface on freeswitch? Or do you think it's a good idea? Or not? Since it is an actual standard, I guess there might be a market for application service providers. Remko ___ 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] javascript: session.recordFile
How? On Tue, Apr 21, 2009 at 8:02 PM, Stephen Crosby stevecr...@gmail.comwrote: You can make a really close estimation based on the size of the recorded file. That's what I've been doing. --Stephen On Tue, Apr 21, 2009 at 10:58 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: Yes, to be able to determine if it's too short or not. On Tue, Apr 21, 2009 at 7:01 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: I'm using session.recordFile in a javascript. How do I check the length of the recorded file? You mean once the file is recorded you want to know how many minutes/seconds long it is? -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] javascript: session.recordFile
You can do a timestamp comparison, but over some long recorded sessions (30+ minutes), I've run into some type of drift in the past that made this method inaccurate enough for me to not use. --Stephen On Tue, Apr 21, 2009 at 11:06 AM, Michael Collins m...@freeswitch.orgwrote: Just curious - could you not set a variable with the value of getTime() right before recording starts and then compare that to getTime() as soon as the recording ends? -MC On Tue, Apr 21, 2009 at 10:58 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: Yes, to be able to determine if it's too short or not. On Tue, Apr 21, 2009 at 7:01 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: I'm using session.recordFile in a javascript. How do I check the length of the recorded file? You mean once the file is recorded you want to know how many minutes/seconds long it is? -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] VoiceXML
Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. I don't know went demand will pick up for VXML on FS but one issue that I think will block it is how robust the FS embedding of pocketsphinx is to the harsh noisy enviroment of telephony. Currently, it lacks this robustness compared to vendor products but once that happens then more will want to use it and VXML will become more in demand? ... hopefully. -Original Message- From: Remko Kloosterman r.klooster...@mtel.nl To: freeswitch-users@lists.freeswitch.org Sent: Tue, 21 Apr 2009 10:57 am Subject: [Freeswitch-users] VoiceXML Hi everyone, ? Did someone ever try and implement a VXML interface on freeswitch? Or do you think it's a good idea? Or not? Since it is an actual standard, I guess there might be a market for application service providers. ? Remko ___ 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] ekiga and freeswitch
just wanted to post this problem as resolved. problem was with ekiga, i needed celt 5.1 on ekiga machine. simple mistake that could easily be avoided if documentation existed. 2009/4/16 Brian West br...@freeswitch.org: I think this is a bug in Ekiga, We do CELT on 114 and Ekiga does Speex on 114 I suspect your client is sending speex frames on 114 instead of celt frames. /b On Apr 16, 2009, at 1:05 PM, e schmidbauer wrote: i've posted the freeswitch svn trunk console debug and sip trace to the pastebin. On Sun, Apr 12, 2009 at 1:34 PM, Brian West br...@freeswitch.org wrote: Collect a full sip trace and FULL console debug. Put it on our pastebin... Chances are Ekiga is doing something stupid... it usually does silly things. Also are you on SVN trunk? /b On Apr 12, 2009, at 12:11 PM, e schmidbauer wrote: Not sure if this is a bug in the program or just in my setup. I've tried using the svn version of freeswitch (as of yesterday) and i got the exact same error. Any input would be appreciated. Thanks! ___ 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 Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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] Adding Spanish support to say
Any updates on this? My offer still stands but for the time being I just need some basic say support for digits... On Wed, Apr 15, 2009 at 4:51 PM, Michael Collins m...@freeswitch.org wrote: FYI, we have translation of the phrase file happening right now. But KK's question is still valid: what does he need to do to get over the hump? -MC -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] Adding Spanish support to say
As soon as you sign up for Cluecon I'll have something for your. ;) Okay, seriously we have been working on this. Let me check in with a few people and see where they're at. For the sake of not having any miscommunications could you please recap what you're doing and the issue you're having with the say support in espanol? Thanks! -MC On Tue, Apr 21, 2009 at 11:47 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Any updates on this? My offer still stands but for the time being I just need some basic say support for digits... On Wed, Apr 15, 2009 at 4:51 PM, Michael Collins m...@freeswitch.org wrote: FYI, we have translation of the phrase file happening right now. But KK's question is still valid: what does he need to do to get over the hump? -MC -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 ___ 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] javascript: session.recordFile
update to trunk and you should have record_ms and record_samples chanvars also playback_ms and playback_samples for playing On Tue, Apr 21, 2009 at 12:58 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote: Yes, to be able to determine if it's too short or not. On Tue, Apr 21, 2009 at 7:01 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: I'm using session.recordFile in a javascript. How do I check the length of the recorded file? You mean once the file is recorded you want to know how many minutes/seconds long it is? -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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] Adding Spanish support to say
I'm still figuring out my schedule! I should know soon...
Here is the description of what I am trying to do from my original post:
I'm trying to add Spanish support to say. I'm using something like:
include
language name=es sound-path=$${base_dir}/sounds/es/mx/asterisk
tts-engine=cepstral tts-voice=callie
X-PRE-PROCESS cmd=include data=demo/*.xml/ !-- Note: this
now grabs whole subdir, previously grabbed only demo.xml --
!--voicemail_en_tts is purely implemented with tts, we have the
files based one that is the default. --
X-PRE-PROCESS cmd=include data=vm/sounds.xml/ !--
vm/tts.xml if you want to use tts and have cepstral --
/language
/include
in conf/lang/es which is included by freeswitch.conf:
X-PRE-PROCESS cmd=include data=lang/es/*.xml/
..right after English. Yet I continue to get
[ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]!
Whenever trying to use say:
action application=say data=es name_spelled pronounced
${caller_id_number}/
What am I missing?
On Tue, Apr 21, 2009 at 3:14 PM, Michael Collins m...@freeswitch.org wrote:
As soon as you sign up for Cluecon I'll have something for your. ;)
Okay, seriously we have been working on this. Let me check in with a few
people and see where they're at. For the sake of not having any
miscommunications could you please recap what you're doing and the issue
you're having with the say support in espanol? Thanks!
-MC
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
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Re: [Freeswitch-users] Adding Spanish support to say
On Tue, Apr 21, 2009 at 12:55 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
I'm still figuring out my schedule! I should know soon...
Hehe, I'm just giving you a hard time. However, we definitely want you at
Cluecon.
Here is the description of what I am trying to do from my original post:
I'm trying to add Spanish support to say. I'm using something like:
include
language name=es sound-path=$${base_dir}/sounds/es/mx/asterisk
tts-engine=cepstral tts-voice=callie
X-PRE-PROCESS cmd=include data=demo/*.xml/ !-- Note: this
now grabs whole subdir, previously grabbed only demo.xml --
!--voicemail_en_tts is purely implemented with tts, we have the
files based one that is the default. --
X-PRE-PROCESS cmd=include data=vm/sounds.xml/ !--
vm/tts.xml if you want to use tts and have cepstral --
/language
/include
in conf/lang/es which is included by freeswitch.conf:
X-PRE-PROCESS cmd=include data=lang/es/*.xml/
..right after English. Yet I continue to get
[ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]!
Whenever trying to use say:
action application=say data=es name_spelled pronounced
${caller_id_number}/
What am I missing?
Okay, dumb questions...
#1 - did you enable mod_say_es in modules.conf and compile it?
#2 - did you load mod_say_es in modules.conf.xml?
If I already asked those questions then my apologies
-MC
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Re: [Freeswitch-users] ACL not working
Nope, I just wanted to allow 1 ip, 192.168.0.100. Diego On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.org wrote: Do you want to allow these IP ranges? /b On Apr 21, 2009, at 6:08 AM, Diego Viola wrote: node type=deny cidr=192.168.0.100/32/ node type=deny cidr=192.168.0.0/24/ Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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] Adding Spanish support to say
On Tue, Apr 21, 2009 at 4:02 PM, Michael Collins m...@freeswitch.org wrote: Okay, dumb questions... #1 - did you enable mod_say_es in modules.conf and compile it? #2 - did you load mod_say_es in modules.conf.xml? If I already asked those questions then my apologies -MC MC, Not dumb questions at all! That was it, at least for the error I was getting. Now say doesn't complain about not knowing es. However, the digits still don't play in Spanish. I wonder if my lang files are correct? Everything there seems pretty straightforward... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
[Freeswitch-users] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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] Errors while installing FreeSwitch 1.0.4pre4.
Don't compile any of the ldap modules. It's not FreeSWITCH fails to build its the OpenLDAP library that is failing. I don't think the ldap modules do much of anything right now in the first place. /b On Apr 21, 2009, at 3:30 PM, technologyinspired wrote: I have found erros while installing FreeSwitch 1.0.4pre4. I am using Toshiba Laptop with FC10. here is the partial output of make command: ... ... Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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
[Freeswitch-users] Errors while installing FreeSwitch 1.0.4pre4.
I have found erros while installing FreeSwitch 1.0.4pre4. I am using Toshiba Laptop with FC10. here is the partial output of make command: ... ... Making servers/slapd/backends.c Add config ... Add ldif ... Making servers/slapd/overlays/statover.c Please run make depend to build dependencies Making all in /usr/src/freeswitch-1.0.4pre4/libs/openldap-2.4.11 Entering subdirectory include Making ldap_config.h Entering subdirectory libraries Making all in /usr/src/freeswitch-1.0.4pre4/libs/openldap-2.4.11/libraries Entering subdirectory liblutil getpeereid.c: In function ‘lutil_getpeereid’: getpeereid.c:65: error: storage size of ‘peercred’ isn’t known make[8]: *** [getpeereid.o] Error 1 make[7]: *** [all-common] Error 1 make[6]: *** [all-common] Error 1 make[5]: *** [/usr/src/freeswitch-1.0.4pre4/libs/openldap-2.4.11/libraries/libldap_r/ libldap_r.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [r...@localhost freeswitch-1.0.4pre4] Even though it is showing that FreeSwitch has been successfully buit and run make install. at the end it gives 2 errors. So is the make process complete or there is some bug? Regards, Vin ___ 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] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
You can use ESL (Event Socket Library) to talk to freeswitch and do the same thing as fastagi. Check out http://wiki.freeswitch.org/wiki/Esl (it builds a native python module too!) Mathieu On 21-Apr-09, at 3:13 PM, technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
On Tue, Apr 21, 2009 at 5:47 PM, Mathieu Rene mrene_li...@avgs.ca wrote: You can use ESL (Event Socket Library) to talk to freeswitch and do the same thing as fastagi. Check out http://wiki.freeswitch.org/wiki/Esl (it builds a native python module too!) There's also the eventsocket protocol for twisted, which allows one to create inbound and outbound socket apps within the same code: http://code.google.com/p/eventsocket Mathieu On 21-Apr-09, at 3:13 PM, technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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 -- Arnaldo M Pereira ___ 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] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
I'm not sure what is meant by this. It's been ages since I did anything with Asterisk so I grow senile. But shouldn't you just be able to use django as your framework and bring the FS functionality into your specific app (or into django if django supports modules) ? One possibility is for your web app to interface with FS's event socket or similar method. Sorry I can't be of much more assistance... -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
try http://wiki.freeswitch.org/wiki/Mod_event_socket On Tue, Apr 21, 2009 at 2:13 PM, technologyinspired technologyinspi...@gmail.com wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] Experience with libpri?
I recently converted our last Asterisk system to FreeSWITCH about a week and a half ago. I have an A101D card on the machine which was run as zaptel for Asterisk. I am currently using wanpipe 3.3.16 in TDM API mode (no more need for 1000 jiffies a second) and libpri 1.4.9 on my system. My PRI is NI2 cpe. I have experienced an issue which (to the best of my knowledge) is due to the wanpipe drivers in TDM API mode and not specific to libpri: I have lost my D-channel twice so far. Otherwise, the system works better than Asterisk ever did. I'm just waiting for the fix from Sangoma. -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.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] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
On Apr 21, 2009, at 4:13 PM, technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? ESL is the answer to your question. Take a look at the wiki. There is a Python binding for it as well. The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Ok, I don't see a problem there ... Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Carefull with Django + FS. You could run over the problem that, since FS is multi domain, you might want to have a SaaS sometime and Django does not connect to multiple DBs with its native ORM. What kind of pointers are you looking for? Thanks, Regards, Vin ___ 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 Jmesquita ___ 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] getting statistics from core db through esl interface
Are you going to donate the complete module to the collectd project? /b On Apr 21, 2009, at 8:52 AM, Leon de Rooij wrote: Thanks, I know that's possible, but I want to get the amount of channels per profile (and even seperated inbound/outbound), not a total of all channels on all profiles combined.. regards, Leon Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] ACL not working
On Tue, Apr 21, 2009 at 1:15 PM, Diego Viola diego.vi...@gmail.com wrote: Nope, I just wanted to allow 1 ip, 192.168.0.100. Then why have a deny for this address? Don't you want something like this? node type=allow cidr=192.168.0.100/32/ -MC Diego On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.org wrote: Do you want to allow these IP ranges? /b On Apr 21, 2009, at 6:08 AM, Diego Viola wrote: node type=deny cidr=192.168.0.100/32/ node type=deny cidr=192.168.0.0/24/ Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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] Adding Spanish support to say
On Tue, Apr 21, 2009 at 1:33 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: On Tue, Apr 21, 2009 at 4:02 PM, Michael Collins m...@freeswitch.org wrote: Okay, dumb questions... #1 - did you enable mod_say_es in modules.conf and compile it? #2 - did you load mod_say_es in modules.conf.xml? If I already asked those questions then my apologies -MC MC, Not dumb questions at all! That was it, at least for the error I was getting. Now say doesn't complain about not knowing es. However, the digits still don't play in Spanish. I wonder if my lang files are correct? Everything there seems pretty straightforward... Gimme about 30 minutes to lab this up and see if I can duplicate your issue... -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
Re: [Freeswitch-users] [CentOS] Modifying init script?
Those files in tree go with the rpm build with goes correctly to those
directories.
Mike
On Apr 21, 2009, at 4:06 AM, Fred-145 wrote:
Hello
I'm installing Freeswitch on a CentOS 5.3 test host. I noticed that
the
freeswitch.init.redhat contains paths that don't match a stock CentOS
install.
All files are currently installed/owned by root: Should I go ahead,
create a
freeswitch user/group, and chown everything under /usr/local/
freeswitch/,
or are there issues if some files aren't owned by root?
Here's the original file, and mine:
= ORIG /usr/src/freeswitch/build/freeswitch.init.redhat
=
PROG_NAME=freeswitch
PID_FILE=${PID_FILE-/opt/freeswitch/log/freeswitch.pid}
FS_USER=${FS_USER-freeswitch}
FS_FILE=${FS_FILE-/opt/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/opt/freeswitch}
LOCK_FILE=/var/lock/subsys/freeswitch
FREESWITCH_ARGS=-nc
RETVAL=0
= My /etc/init.d/freeswitch =
PROG_NAME=freeswitch
PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid}
#All files installed/owned by root: Should I go ahead and create a
freeswitch user anyway?
FS_USER=${FS_USER-freeswitch}
FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/usr/local/freeswitch}
LOCK_FILE=/var/lock/subsys/freeswitch
FREESWITCH_ARGS=-nc
RETVAL=0
==
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Re: [Freeswitch-users] Help with mod_managed under Windows
Could you translate these into english? On Apr 21, 2009, at 7:08 AM, Guido Kuth wrote: I am playing around with FS (Windows) for one month now. First I tried using FreeSwitch.NET which is a good class library for inbound event socket. Unfortunatley it can't be used for outbound event socket. So I read the wiki back ond forth and also searched the net and found that I should use mod_managed. So I downloaded mod_managed in source from svn and compiled it with C# 2008 Express Edition. After that I got a dll. FreeSwitch.Managed.dll and copied it to the mod dir of FS. The Problem is that I get and error when FS loads mod_managed and I don't know what I should do with that. 2009-04-21 11:26:44 [INFO] mod_managed.cpp:314 mod_managed_load() Loading mod_ma naged (Common Language Infrastructure), Microsoft CLR Version 2009-04-21 11:26:44 [ERR] mod_managed.cpp:333 mod_managed_load() Load did not re turn true. System.Reflection.TargetInvocationException: Ein Aufrufziel hat einen Ausnahmefehler verursacht. --- System.TypeInitializationException: Der Typenin itialisierer f³r FreeSWITCH.Native.freeswitch hat eine Ausnahme verursacht. -- - System.EntryPointNotFoundException: Der Einstiegspunkt CSharp_SWITCH_READ_TE RMINATOR_USED_VARIABLE_get wurde nicht in der DLL mod_managed gefunden. bei FreeSWITCH .Native.freeswitchPINVOKE.SWITCH_READ_TERMINATOR_USED_VARIABLE_ get() bei FreeSWITCH.Native.freeswitch..cctor() --- Ende der internen Ausnahmestapel³berwachung --- bei FreeSWITCH.Native.freeswitch.get_SWITCH_GLOBAL_dirs() bei FreeSWITCH.Loader.Load() --- Ende der internen Ausnahmestapel³berwachung --- bei System.RuntimeMethodHandle._InvokeMethodFast(Object target, Object[] argu ments, SignatureStruct sig, MethodAttributes methodAttributes, RuntimeTypeHandl e typeOwner) bei System.RuntimeMethodHandle.InvokeMethodFast(Object target, Object[] argum ents, Signature sig, MethodAttributes methodAttributes, RuntimeTypeHandle typeOw ner) bei System.Reflection.RuntimeMethodInfo.Invoke(Object obj, BindingFlags invok eAttr, Binder binder, Object[] parameters, CultureInfo culture, Boolean skipVisi bilityChecks) bei System.Reflection.RuntimeMethodInfo.Invoke(Object obj, BindingFlags invok eAttr, Binder binder, Object[] parameters, CultureInfo culture) bei mod_managed_load(switch_loadable_module_interface** module_interface, apr _pool_t* pool) 2009-04-21 11:26:44 [CRIT] switch_loadable_module.c:845 switch_loadable_module_l oad_file() Error Loading module C:\Programme\FreeSWITCH\mod \mod_managed.dll **Module load routine returned an error** Please help me with that. Thanks...Guido ___ 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] VIA messages?
http://www.ietf.org/rfc/rfc3261.txt 8.1.1.7 Via On Apr 21, 2009, at 10:48 AM, Fred-145 wrote: Hello I'm reading the examples, and found one on how to use a Sipura SPA-2000: http://wiki.freeswitch.org/wiki/Sipura_STUN I was wondering what the VIA messages mean, and whether I should enable those when the unit is used behind a NAT router. I thought Freeswitch would take care of NAT, but it looks like SIP devices have to be configured for this. Anybody knows what VIA messages are for? ___ 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] VoiceXML
FreeSWITCH has a generic ASR interface to plug into any asr engine. The quality of the integrated free asr has little to do with VXML. If you want more robust asr, you will likely need to pay for it. Mike On Apr 21, 2009, at 2:35 PM, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. I don't know went demand will pick up for VXML on FS but one issue that I think will block it is how robust the FS embedding of pocketsphinx is to the harsh noisy enviroment of telephony. Currently, it lacks this robustness compared to vendor products but once that happens then more will want to use it and VXML will become more in demand ... hopefully. ___ 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] VoiceXML
On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave ___ 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] ACL not working
I was just trying to deny everything, and I got confused at what the default in the list made, but I got it now. So I have list name=domains default=deny and that alone denies the registration, which is what I want, but I can still make calls. And I have this: param name=apply-inbound-acl value=domains/ Shouldn't the domains which is defaulted to deny block the inbound calls? Thanks, I hope this doesn't make anyone nervous, just trying to learn :) Regards, Diego On Tue, Apr 21, 2009 at 5:34 PM, Michael Collins m...@freeswitch.org wrote: On Tue, Apr 21, 2009 at 1:15 PM, Diego Viola diego.vi...@gmail.comwrote: Nope, I just wanted to allow 1 ip, 192.168.0.100. Then why have a deny for this address? Don't you want something like this? node type=allow cidr=192.168.0.100/32/ -MC Diego On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.org wrote: Do you want to allow these IP ranges? /b On Apr 21, 2009, at 6:08 AM, Diego Viola wrote: node type=deny cidr=192.168.0.100/32/ node type=deny cidr=192.168.0.0/24/ Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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] Adding Spanish support to say
sound_prefix? Mike On Apr 21, 2009, at 8:02 PM, Michael Collins wrote: Kristian, The symptom I'm experiencing is that no matter what language I specify, it still plays the English sound files. Is that what you're experiencing? I've run it with debug logging turned on and combed through the source code and I can't find anything that explicitly falls back to English when the other language has failed for some reason. In fact, I get no errors of any kind. I'll have to defer to the masters on this one. :) -MC On Tue, Apr 21, 2009 at 2:40 PM, Michael Collins m...@freeswitch.org wrote: On Tue, Apr 21, 2009 at 1:33 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: On Tue, Apr 21, 2009 at 4:02 PM, Michael Collins m...@freeswitch.org wrote: Okay, dumb questions... #1 - did you enable mod_say_es in modules.conf and compile it? #2 - did you load mod_say_es in modules.conf.xml? If I already asked those questions then my apologies -MC MC, Not dumb questions at all! That was it, at least for the error I was getting. Now say doesn't complain about not knowing es. However, the digits still don't play in Spanish. I wonder if my lang files are correct? Everything there seems pretty straightforward... Gimme about 30 minutes to lab this up and see if I can duplicate your issue... ___ 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] ACL not working
This alone should be able to block inbound calls right?:
internal.xml:
param name=apply-inbound-acl value=domains/
acl.conf.xml:
list name=domains default=deny
node type=allow domain=$${domain}/
/list
vars.xml:
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
On Tue, Apr 21, 2009 at 8:04 PM, Diego Viola diego.vi...@gmail.com wrote:
I was just trying to deny everything, and I got confused at what the
default in the list made, but I got it now.
So I have list name=domains default=deny and that alone denies
the registration, which is what I want, but I can still make calls.
And I have this: param name=apply-inbound-acl value=domains/
Shouldn't the domains which is defaulted to deny block the inbound
calls?
Thanks, I hope this doesn't make anyone nervous, just trying to learn :)
Regards,
Diego
On Tue, Apr 21, 2009 at 5:34 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Apr 21, 2009 at 1:15 PM, Diego Viola diego.vi...@gmail.comwrote:
Nope, I just wanted to allow 1 ip, 192.168.0.100.
Then why have a deny for this address? Don't you want something like this?
node type=allow cidr=192.168.0.100/32/
-MC
Diego
On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.orgwrote:
Do you want to allow these IP ranges?
/b
On Apr 21, 2009, at 6:08 AM, Diego Viola wrote:
node type=deny cidr=192.168.0.100/32/
node type=deny cidr=192.168.0.0/24/
Brian West
br...@freeswitch.org
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Re: [Freeswitch-users] ACL not working
If I turn internal_auth_calls to false it blocks... but why I can't do it
with internal_auth_calls=true?
On Tue, Apr 21, 2009 at 8:45 PM, Diego Viola diego.vi...@gmail.com wrote:
This alone should be able to block inbound calls right?:
internal.xml:
param name=apply-inbound-acl value=domains/
acl.conf.xml:
list name=domains default=deny
node type=allow domain=$${domain}/
/list
vars.xml:
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
On Tue, Apr 21, 2009 at 8:04 PM, Diego Viola diego.vi...@gmail.comwrote:
I was just trying to deny everything, and I got confused at what the
default in the list made, but I got it now.
So I have list name=domains default=deny and that alone denies
the registration, which is what I want, but I can still make calls.
And I have this: param name=apply-inbound-acl value=domains/
Shouldn't the domains which is defaulted to deny block the inbound
calls?
Thanks, I hope this doesn't make anyone nervous, just trying to learn :)
Regards,
Diego
On Tue, Apr 21, 2009 at 5:34 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Apr 21, 2009 at 1:15 PM, Diego Viola diego.vi...@gmail.comwrote:
Nope, I just wanted to allow 1 ip, 192.168.0.100.
Then why have a deny for this address? Don't you want something like
this?
node type=allow cidr=192.168.0.100/32/
-MC
Diego
On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.orgwrote:
Do you want to allow these IP ranges?
/b
On Apr 21, 2009, at 6:08 AM, Diego Viola wrote:
node type=deny cidr=192.168.0.100/32/
node type=deny cidr=192.168.0.0/24/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] ACL not working
I'm trying to block inbound calls with internal_auth_calls=true.
On Tue, Apr 21, 2009 at 8:46 PM, Diego Viola diego.vi...@gmail.com wrote:
If I turn internal_auth_calls to false it blocks... but why I can't do it
with internal_auth_calls=true?
On Tue, Apr 21, 2009 at 8:45 PM, Diego Viola diego.vi...@gmail.comwrote:
This alone should be able to block inbound calls right?:
internal.xml:
param name=apply-inbound-acl value=domains/
acl.conf.xml:
list name=domains default=deny
node type=allow domain=$${domain}/
/list
vars.xml:
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
On Tue, Apr 21, 2009 at 8:04 PM, Diego Viola diego.vi...@gmail.comwrote:
I was just trying to deny everything, and I got confused at what the
default in the list made, but I got it now.
So I have list name=domains default=deny and that alone denies
the registration, which is what I want, but I can still make calls.
And I have this: param name=apply-inbound-acl value=domains/
Shouldn't the domains which is defaulted to deny block the inbound
calls?
Thanks, I hope this doesn't make anyone nervous, just trying to learn :)
Regards,
Diego
On Tue, Apr 21, 2009 at 5:34 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Apr 21, 2009 at 1:15 PM, Diego Viola diego.vi...@gmail.comwrote:
Nope, I just wanted to allow 1 ip, 192.168.0.100.
Then why have a deny for this address? Don't you want something like
this?
node type=allow cidr=192.168.0.100/32/
-MC
Diego
On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.orgwrote:
Do you want to allow these IP ranges?
/b
On Apr 21, 2009, at 6:08 AM, Diego Viola wrote:
node type=deny cidr=192.168.0.100/32/
node type=deny cidr=192.168.0.0/24/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] ACL not working
I got it, thanks people :D
On Tue, Apr 21, 2009 at 8:57 PM, Diego Viola diego.vi...@gmail.com wrote:
I'm trying to block inbound calls with internal_auth_calls=true.
On Tue, Apr 21, 2009 at 8:46 PM, Diego Viola diego.vi...@gmail.comwrote:
If I turn internal_auth_calls to false it blocks... but why I can't do it
with internal_auth_calls=true?
On Tue, Apr 21, 2009 at 8:45 PM, Diego Viola diego.vi...@gmail.comwrote:
This alone should be able to block inbound calls right?:
internal.xml:
param name=apply-inbound-acl value=domains/
acl.conf.xml:
list name=domains default=deny
node type=allow domain=$${domain}/
/list
vars.xml:
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
On Tue, Apr 21, 2009 at 8:04 PM, Diego Viola diego.vi...@gmail.comwrote:
I was just trying to deny everything, and I got confused at what the
default in the list made, but I got it now.
So I have list name=domains default=deny and that alone denies
the registration, which is what I want, but I can still make calls.
And I have this: param name=apply-inbound-acl value=domains/
Shouldn't the domains which is defaulted to deny block the inbound
calls?
Thanks, I hope this doesn't make anyone nervous, just trying to learn :)
Regards,
Diego
On Tue, Apr 21, 2009 at 5:34 PM, Michael Collins
m...@freeswitch.orgwrote:
On Tue, Apr 21, 2009 at 1:15 PM, Diego Viola diego.vi...@gmail.comwrote:
Nope, I just wanted to allow 1 ip, 192.168.0.100.
Then why have a deny for this address? Don't you want something like
this?
node type=allow cidr=192.168.0.100/32/
-MC
Diego
On Tue, Apr 21, 2009 at 9:27 AM, Brian West br...@freeswitch.orgwrote:
Do you want to allow these IP ranges?
/b
On Apr 21, 2009, at 6:08 AM, Diego Viola wrote:
node type=deny cidr=192.168.0.100/32/
node type=deny cidr=192.168.0.0/24/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] ACL not working
Diego, I highly recommend you seek professional help... You seem to be talking to yourself A LOT! :P just kidding... good you understand it now! /b On Apr 21, 2009, at 8:44 PM, Diego Viola wrote: I got it, thanks people :D Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] VoiceXML
I was initially turned off by VXML when it came out because the first way they tried to make money off it was to sell the clients that let you build the actual xml. I was not really motivated to pay money to be able to just generate xml just so i could code a free server for it so I lost interest. I did hear there is now finally a free one out there. so that may make it a little more reasonable. I've commented in the past that I'm totally open to supporting VXML but we have never had the public interest, time or resources to work on it thus far. On Tue, Apr 21, 2009 at 6:15 PM, David Knell d...@3c.co.uk wrote: On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] ACL not working
Thanks Brian. Couldn't have made it without your help. Regards, Diego On Tue, Apr 21, 2009 at 9:55 PM, Brian West br...@freeswitch.org wrote: Diego, I highly recommend you seek professional help... You seem to be talking to yourself A LOT! :P just kidding... good you understand it now! /b On Apr 21, 2009, at 8:44 PM, Diego Viola wrote: I got it, thanks people :D Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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] VoiceXML
Directly in vxml. -Original Message- From: David Knell d...@3c.co.uk To: freeswitch-users@lists.freeswitch.org Sent: Tue, 21 Apr 2009 4:15 pm Subject: Re: [Freeswitch-users] VoiceXML On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave ___ 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] ACL not working
Diego Viola diego.vi...@gmail.com wrote: I got it, thanks people :D Could you now add it to the documentation? ___ 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] VoiceXML
Isn't ASR optional in VXML? /b On Apr 21, 2009, at 6:15 PM, David Knell wrote: Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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
[Freeswitch-users] Recomended tools and workflow used for prompt recording
Hi, I know this is probably slightly offtopic for this list and if so please point me to a good place to ask. Thanks. When recording prompts, what is the recommended way to do so? All in one file and then split up as a post process? How do you make sure the prompts have the same number of ms of silence at the beginning and end of each clip? If you have any tips and or links I would be very interested in what you have to say. Thank you -- Sent from my mobile device ___ 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] ACL not working
Sure. On Tue, Apr 21, 2009 at 10:37 PM, Jason White ja...@jasonjgw.net wrote: Diego Viola diego.vi...@gmail.com wrote: I got it, thanks people :D Could you now add it to the documentation? ___ 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] Recomended tools and workflow used for prompt recording
You have to do it all post process. /b On Apr 21, 2009, at 9:42 PM, Mike Fedyk wrote: Hi, I know this is probably slightly offtopic for this list and if so please point me to a good place to ask. Thanks. When recording prompts, what is the recommended way to do so? All in one file and then split up as a post process? How do you make sure the prompts have the same number of ms of silence at the beginning and end of each clip? If you have any tips and or links I would be very interested in what you have to say. Thank you Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Recomended tools and workflow used for prompt recording
Do you do anything to help automate the process? Sound markers that can be detected by post processing tools like sox or audacity? I've done this once before with audacity using labels and multi-file exports and it wasn't a quick process. Is there a better way or some way to script some of it? On 4/21/09, Brian West br...@freeswitch.org wrote: You have to do it all post process. /b On Apr 21, 2009, at 9:42 PM, Mike Fedyk wrote: Hi, I know this is probably slightly offtopic for this list and if so please point me to a good place to ask. Thanks. When recording prompts, what is the recommended way to do so? All in one file and then split up as a post process? How do you make sure the prompts have the same number of ms of silence at the beginning and end of each clip? If you have any tips and or links I would be very interested in what you have to say. Thank you Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -- Sent from my mobile device ___ 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] How to tell if 100 Trying received
Hi Anthony,
Thanks for the reply!
While waiting for my question to appear on the list yesterday (6H delay at
yoda.ostag.org... is first post moderated?) I went deep into the SIP stack
and figured out the solution: You just have to give NTATAG_PASS_100(1) as
one of the tags for nua_create. Then you get a sofia event for it. I guess
the author has made it easier since your last discussion.
I have changed my mod_sofia to do this. I also added a channel flag which
is set if any response has been received from the remote end (be it 100,
18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to
time-out a call quickly.
Would you/anyone be interested in a patch to do this? If so please let me
know the procedure for posting patches etc.
{P^/
On Tue, 21 Apr 2009 at 10:03 -0500, Anthony Minessale wrote:
That 100 trying is handled deep in the sip stack.
The author of sofia said it would be a big job to bring that up to the even
callback.
Someone may be able to persuade him to allow you to pass a global timeout
waiting for 100
or something but no solution exists atm
On Tue, Apr 21, 2009 at 3:32 AM, John Dalgliesh jo...@defyne.org wrote:
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of
gateways that can make the call. However, if one of them is down or has
some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it
hasn't been received in a couple of seconds then go on and try the next
gateway.
But I can't find a flag/event/state which corresponds to receipt of this
message. Can anyone tell me where I should be looking? I put a debug print
in sofia_event_callback for every event but there doesn't seem to be one
fired for this condition.
Thanks in advance.
{P^/
John
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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 34, Issue 123
, at 3:13 PM, technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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 -- Arnaldo M Pereira -- next part -- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090421/60f65aaa/attachment-0001.html -- Message: 3 Date: Tue, 21 Apr 2009 15:51:46 -0500 From: Yossi Neiman freeswi...@cartissolutions.com Subject: Re: [Freeswitch-users] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc To: freeswitch-users@lists.freeswitch.org Message-ID: 49ee31e2.10...@cartissolutions.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed I'm not sure what is meant by this. It's been ages since I did anything with Asterisk so I grow senile. But shouldn't you just be able to use django as your framework and bring the FS functionality into your specific app (or into django if django supports modules) ? One possibility is for your web app to interface with FS's event socket or similar method. Sorry I can't be of much more assistance... -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin ___ 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 -- Message: 4 Date: Tue, 21 Apr 2009 15:55:42 -0500 From: Yossi Neiman freeswi...@cartissolutions.com Subject: Re: [Freeswitch-users] Experience with libpri? To: freeswitch-users@lists.freeswitch.org Message-ID: 49ee32ce.4030...@cartissolutions.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed I recently converted our last Asterisk system to FreeSWITCH about a week and a half ago. I have an A101D card on the machine which was run as zaptel for Asterisk. I am currently using wanpipe 3.3.16 in TDM API mode (no more need for 1000 jiffies a second) and libpri 1.4.9 on my system. My PRI is NI2 cpe. I have experienced an issue which (to the best of my knowledge) is due to the wanpipe drivers in TDM API mode and not specific to libpri: I have lost my D-channel twice so far. Otherwise, the system works better than Asterisk ever did. I'm just waiting for the fix from Sangoma. -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com -- Message: 5 Date: Tue, 21 Apr 2009 15:55:50 -0500 From: Anthony Minessale anthony.miness...@gmail.com Subject: Re: [Freeswitch-users] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc To: freeswitch-users@lists.freeswitch.org Message-ID: 191c3a030904211355q31416fak2f6007c2654e0...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 try http://wiki.freeswitch.org/wiki
Re: [Freeswitch-users] How to tell if 100 Trying received
Patches should be posted to http://jira.freeswitch.org,
Thanks for digging in.
/b
On Apr 21, 2009, at 10:22 PM, John Dalgliesh wrote:
Hi Anthony,
Thanks for the reply!
While waiting for my question to appear on the list yesterday (6H
delay at
yoda.ostag.org... is first post moderated?) I went deep into the SIP
stack
and figured out the solution: You just have to give
NTATAG_PASS_100(1) as
one of the tags for nua_create. Then you get a sofia event for it. I
guess
the author has made it easier since your last discussion.
I have changed my mod_sofia to do this. I also added a channel flag
which
is set if any response has been received from the remote end (be it
100,
18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to
time-out a call quickly.
Would you/anyone be interested in a patch to do this? If so please
let me
know the procedure for posting patches etc.
{P^/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] How to tell if 100 Trying received
I guess maybe he added it as a result of our last conversation and never
told me.
yes, if you do it elegantly we can add that patch.
submit what you have to jira in we can do code review.
On Tue, Apr 21, 2009 at 10:38 PM, Brian West br...@freeswitch.org wrote:
Patches should be posted to http://jira.freeswitch.org,
Thanks for digging in.
/b
On Apr 21, 2009, at 10:22 PM, John Dalgliesh wrote:
Hi Anthony,
Thanks for the reply!
While waiting for my question to appear on the list yesterday (6H delay at
yoda.ostag.org... is first post moderated?) I went deep into the SIP stack
and figured out the solution: You just have to give NTATAG_PASS_100(1) as
one of the tags for nua_create. Then you get a sofia event for it. I guess
the author has made it easier since your last discussion.
I have changed my mod_sofia to do this. I also added a channel flag which
is set if any response has been received from the remote end (be it 100,
18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to
time-out a call quickly.
Would you/anyone be interested in a patch to do this? If so please let me
know the procedure for posting patches etc.
{P^/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] javascript: session.recordFile
Thanks! :) On Tue, Apr 21, 2009 at 9:34 PM, Anthony Minessale anthony.miness...@gmail.com wrote: update to trunk and you should have record_ms and record_samples chanvars also playback_ms and playback_samples for playing On Tue, Apr 21, 2009 at 12:58 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote: Yes, to be able to determine if it's too short or not. On Tue, Apr 21, 2009 at 7:01 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Apr 21, 2009 at 4:23 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote: I'm using session.recordFile in a javascript. How do I check the length of the recorded file? You mean once the file is recorded you want to know how many minutes/seconds long it is? -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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] VoiceXML
You can use VXML without ASR, but ASR and TTS are both required parts of the specs. Mike On Apr 21, 2009, at 10:41 PM, Brian West wrote: Isn't ASR optional in VXML? /b On Apr 21, 2009, at 6:15 PM, David Knell wrote: Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? ___ 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
