[Freeswitch-users] Gateway issue with no audio
My freeswitch is using public IP. I setup a gateway registering to voipstunt, and put it under internal profile. I tried to make call, and I got no RTP back from the provider... Tried treating NAT issue by changing IP address, internal IP, external IP. But no use, still getting no audio. Finally, I gave up play around with the internal profile and put the gateway *settings under external profile. And magically, it worked.* I am getting audio now. But it leads me to wonders, what's the core difference between external profile and internal profile. Even if I set the external SIP IP and exteranl RTP IP to the public IP in internal profile, I am still getting no audio. Can anyone clear the concept for me here? by the way, I am using freeswitch 1.4 stable version. -- Henry Huang ___ 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] sched_api doesn't get launched
Anthony:
You are da man. Thank you. It's very kind of you to release me from endless
trial :P
On Sat, Oct 24, 2009 at 1:47 AM, Chris Burns ch...@cloudtel.com wrote:
poor bbhenry :)
Added in r15207 please test and update docu if necessary:
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
On October 23, 2009 11:02:07 am Anthony Minessale wrote:
it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer variable to
trunk.
On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang
red.rain.se...@gmail.comwrote:
Thanks to c6burns on IRC channel for the tip to use execute_on_answer
in
combination with eval, and of course everyone here that pointing me to
the right direction.
I was able to execute sched_api with eval, but not with the combination
of execute_on_answer. The argument just don't get parsed as the eval
argument. Here is the code:
setVariable(execute_on_answer, eval sched_api(+%ld none
uuid_displace
some uuid start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20
mux));
Here is the log:
2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
sched_api_function (none) to run at 1256285888
EXECUTE sofia/internal/1688...@192.168.1.66 *eval(+OK *Added: 2
)
2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
1688...@192.168.1.66 *execute on answer: eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
EXECUTE sofia/internal/1688...@192.168.1.66 eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
The first part of the log is when I just do execute(eval,
${sched_api(+%ld none uuid_displace some uuid start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20
mux)});
And I simple get a OK, and then later on the sched_api gets executed.
However, on the excerpt of the second log. which I use the combination
with execute_on_answer like specified in the yellow hight code above.
It
doesn't simply return a OK, instead it shows the whole blob of code.
and
does nothing later.
On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Yes you need an API object as described in other email.
Which line of code from java caused that segfault
It looks like a simple NULL string issue that we may want to hunt
down.
On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang
red.rain.se...@gmail.comwrote:
I can't seem to find the right thing to use in mod_java to execute
api
commands, only api_after_bridge
2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
sofia/internal/1688...@192.168.1.66!
#
# A fatal error has been detected by the Java Runtime Environment:
#
# SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
#
# JRE version: 6.0_16-b01
# Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86
)
# Problematic frame:
# C [libc.so.6+0x6f480] strcpy+0x10
#
# An error report file with more information is saved as:
# /usr/local/freeswitch/bin/hs_err_pid1927.log
2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
Application sched_api
2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
sofia/internal/1688...@192.168.1.66 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
#
# If you would like to submit a bug report, please visit:
# http://java.sun.com/webapps/bugreport/crash.jsp
# The crash happened outside the Java Virtual Machine in native code.
# See problematic frame for where to report the bug.
On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins
m...@freeswitch.orgwrote:
On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
red.rain.se...@gmail.com
wrote:
So how would you trigger it from a script dialplan? The only time
it
seemed to work is when I did setVariable(api_after_bridge,
sched_api blah blah blah);
but then it gets executed after the channel's been teared down. I
thought api_after_bridge means right after the call gets connected.
I need something to execute an api command right before or right
after the call gets bridged.
api_after_bridge is a channel variable, so using setVariable works
just
fine. If you need to sched_api is an API only. Check these out:
http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
So you need an API object in order to use it. I don't know the
syntax
for creating an api obj in Java but in Lua it goes like this:
api = freeswitch.API();
res = api:execute(sched_api,+300 none my_api my_api_args)
Remember, if the method you are using isn't found in the dial plan
tools then it isn't a dial plan
Re: [Freeswitch-users] sched_api doesn't get launched
Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
combination with eval, and of course everyone here that pointing me to the
right direction.
I was able to execute sched_api with eval, but not with the combination of
execute_on_answer. The argument just don't get parsed as the eval argument.
Here is the code:
setVariable(execute_on_answer, eval sched_api(+%ld none uuid_displace
some uuid start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux));
Here is the log:
2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
sched_api_function (none) to run at 1256285888
EXECUTE sofia/internal/1688...@192.168.1.66 *eval(+OK *Added: 2
)
2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
1688...@192.168.1.66 *execute on answer: eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
EXECUTE sofia/internal/1688...@192.168.1.66 eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
The first part of the log is when I just do execute(eval,
${sched_api(+%ld none uuid_displace some uuid start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)});
And I simple get a OK, and then later on the sched_api gets executed.
However, on the excerpt of the second log. which I use the combination with
execute_on_answer like specified in the yellow hight code above. It doesn't
simply return a OK, instead it shows the whole blob of code. and does
nothing later.
On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Yes you need an API object as described in other email.
Which line of code from java caused that segfault
It looks like a simple NULL string issue that we may want to hunt down.
On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang red.rain.se...@gmail.comwrote:
I can't seem to find the right thing to use in mod_java to execute api
commands, only api_after_bridge
2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
sofia/internal/1688...@192.168.1.66!
#
# A fatal error has been detected by the Java Runtime Environment:
#
# SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
#
# JRE version: 6.0_16-b01
# Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
# Problematic frame:
# C [libc.so.6+0x6f480] strcpy+0x10
#
# An error report file with more information is saved as:
# /usr/local/freeswitch/bin/hs_err_pid1927.log
2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
Application sched_api
2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
sofia/internal/1688...@192.168.1.66 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
#
# If you would like to submit a bug report, please visit:
# http://java.sun.com/webapps/bugreport/crash.jsp
# The crash happened outside the Java Virtual Machine in native code.
# See problematic frame for where to report the bug.
On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.orgwrote:
On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
red.rain.se...@gmail.comwrote:
So how would you trigger it from a script dialplan? The only time it
seemed to work is when I did setVariable(api_after_bridge, sched_api
blah
blah blah);
but then it gets executed after the channel's been teared down. I
thought api_after_bridge means right after the call gets connected.
I need something to execute an api command right before or right after
the call gets bridged.
api_after_bridge is a channel variable, so using setVariable works just
fine. If you need to sched_api is an API only. Check these out:
http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
So you need an API object in order to use it. I don't know the syntax for
creating an api obj in Java but in Lua it goes like this:
api = freeswitch.API();
res = api:execute(sched_api,+300 none my_api my_api_args)
Remember, if the method you are using isn't found in the dial plan tools
then it isn't a dial plan application. Make sure it's on the list:
http://wiki.freeswitch.org/wiki/Mod_dptools
On the other hand, API commands are listed here:
http://wiki.freeswitch.org/wiki/Mod_commands
dptools require a session object, api commands require an api object...
-MC
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Re: [Freeswitch-users] sched_api doesn't get launched
I can't seem to find the right thing to use in mod_java to execute api commands, only api_after_bridge 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1688...@192.168.1.66! # # A fatal error has been detected by the Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624 # # JRE version: 6.0_16-b01 # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 ) # Problematic frame: # C [libc.so.6+0x6f480] strcpy+0x10 # # An error report file with more information is saved as: # /usr/local/freeswitch/bin/hs_err_pid1927.log 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid Application sched_api 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup sofia/internal/1688...@192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # The crash happened outside the Java Virtual Machine in native code. # See problematic frame for where to report the bug. On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.org wrote: On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang red.rain.se...@gmail.comwrote: So how would you trigger it from a script dialplan? The only time it seemed to work is when I did setVariable(api_after_bridge, sched_api blah blah blah); but then it gets executed after the channel's been teared down. I thought api_after_bridge means right after the call gets connected. I need something to execute an api command right before or right after the call gets bridged. api_after_bridge is a channel variable, so using setVariable works just fine. If you need to sched_api is an API only. Check these out: http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) Remember, if the method you are using isn't found in the dial plan tools then it isn't a dial plan application. Make sure it's on the list: http://wiki.freeswitch.org/wiki/Mod_dptools On the other hand, API commands are listed here: http://wiki.freeswitch.org/wiki/Mod_commands dptools require a session object, api commands require an api object... -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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] sched_api doesn't get launched
So how would you trigger it from a script dialplan? The only time it seemed
to work is when I did setVariable(api_after_bridge, sched_api blah blah
blah);
but then it gets executed after the channel's been teared down. I thought
api_after_bridge means right after the call gets connected.
I need something to execute an api command right before or right after the
call gets bridged.
On Fri, Oct 16, 2009 at 9:27 PM, Michael Jerris m...@jerris.com wrote:
sched_api is a fsapi command not a dialplan application, I believe
sched_hangup is both.
Mike
On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute(sched_hangup, +300 alloted_timeout);
However, when I try to run sched_api in the same way, system log returns
that it's an invalid application. I have also tried to trigger it with many
conditional channel variable api calls , but non of them seemed to execute
the api command (because I turned on the highest level of debugging and see
no where the sched_api is being called.
The closest thing I got was by using api_after_bridge like the following,
but it only launches when the bridge is teared down(which is not what I
want). I originally thought after bridge means right after the 2 party is
connected. All I want is to be able to play some message to leg A at certain
time.
setVariable(api_after_bridge, sched_api +10 none uuid_displace ${uuid}
start /path/to/some.wav 20 mux);
I have been struggling with different combination for a week now.. Please
shed some light if you know something.
Thanks,
--
Henry Huang
UniC Solution - Communication Unified
VoIP Open Source software Consultant
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[Freeswitch-users] sched_api doesn't get launched
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute(sched_hangup, +300 alloted_timeout);
However, when I try to run sched_api in the same way, system log returns
that it's an invalid application. I have also tried to trigger it with many
conditional channel variable api calls , but non of them seemed to execute
the api command (because I turned on the highest level of debugging and see
no where the sched_api is being called.
The closest thing I got was by using api_after_bridge like the following,
but it only launches when the bridge is teared down(which is not what I
want). I originally thought after bridge means right after the 2 party is
connected. All I want is to be able to play some message to leg A at certain
time.
setVariable(api_after_bridge, sched_api +10 none uuid_displace ${uuid}
start /path/to/some.wav 20 mux);
I have been struggling with different combination for a week now.. Please
shed some light if you know something.
Thanks,
--
Henry Huang
UniC Solution - Communication Unified
VoIP Open Source software Consultant
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Re: [Freeswitch-users] sched_api doesn't get launched
Diego:
You probably miss understood me. I said I was able to make sched_hangup
work, but not the sched_api in the same way I script for sched_hangup
The problem was on the second paragraph.
thanks,
On Wed, Oct 14, 2009 at 2:45 AM, Diego Viola diego.vi...@gmail.com wrote:
You need to pass the UUID to sched_hangup.
Usage: sched_hangup [+]time uuid [cause]
http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup
On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang b_ball_he...@hotmail.comwrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute(sched_hangup, +300 alloted_timeout);
However, when I try to run sched_api in the same way, system log returns
that it's an invalid application. I have also tried to trigger it with many
conditional channel variable api calls , but non of them seemed to execute
the api command (because I turned on the highest level of debugging and see
no where the sched_api is being called.
The closest thing I got was by using api_after_bridge like the
following, but it only launches when the bridge is teared down(which is not
what I want). I originally thought after bridge means right after the 2
party is connected. All I want is to be able to play some message to leg A
at certain time.
setVariable(api_after_bridge, sched_api +10 none uuid_displace ${uuid}
start /path/to/some.wav 20 mux);
I have been struggling with different combination for a week now.. Please
shed some light if you know something.
Thanks,
--
Henry Huang
UniC Solution - Communication Unified
VoIP Open Source software Consultant
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Re: [Freeswitch-users] Some help with my post-paid billing project
Diego:
Didn't wrote a prepaid system already? how did you compare the prefix in
that system then?
On Wed, Oct 14, 2009 at 10:07 AM, TTNC - Adnan Barakat techni...@ttnc.co.uk
wrote:
Diego Viola wrote:
I'm using MySQL now but I will try PostgreSQL with the prefix module, is
there a way to do that without the prefix module and with regular SQL?
Any examples?
SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
LENGTH(prefix)) LIMIT 1
Adnan
Diego
On Tue, Oct 13, 2009 at 10:45 PM, Even André Fiskvik grev...@me.com
mailto:grev...@me.com wrote:
What database are you using?
You could do this with regular SQL, but it would by a costly
operation,
for PostgreSQL we're using the prefix module:
http://pgfoundry.org/projects/prefix/
You can then match the closest prefix by using something like
WHERE myprefix_col @ caller_destination_number ORDER BY LENGTH
(myprefix_col::text) LIMIT 1;
Best regards,
Even André
On 13. okt. 2009, at 23.53, Diego Viola wrote:
Hello,
I'm trying to write a post-paid billing script, I have the CDR on
my
database and also a rates table, the CDR contains fields like
caller_destination_number, variable_duration, etc. and the rates
table contains: destination, prefix, rate (cost).
The problem is that I can't just strip the destination number to
take the prefix from it because I have to deal with destination
numbers from different countries and they all have different prefix
lengths... so I need to find another way to take the prefix from
the
destination number.
Any ideas how to do this?
Thanks,
Diego
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Re: [Freeswitch-users] how to match '#' in XML dialplan ?
Daqiang: How do you make your IP phone not dial right after you press #? Usually the IP phone will dial the number already once you pushed # On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang wangdq@gmail.com wrote: it's work . Thank you very much . 2009/10/11 Michael Collins m...@freeswitch.org Some characters need a backslash to match in a regular expression. However, # is not one of them. I think your regex is wrong: condition field=destination_number expression=^1#(d+)#(d+)$/ It should probably be: condition field=destination_number expression=^1#(\d+)#(\d+)$/ Note the backslashes in front of the d+ entries. \d means match a digit whereas a bare d means make a lowercase d character. Hope that helps. -MC P.S. - The * character does need to be escaped in regexes. See the default.xml dialplan file for some obvious examples. On Sat, Oct 10, 2009 at 6:24 AM, Milena testeado...@gmail.com wrote: escape character is '\'try condition field=destination_number expression=^1\#(d+)\#(d+)$/ 2009/10/10 daqiang wang wangdq@gmail.com hello every one : I want to match the # in XML dialplan , how to do ? example : 1## . how to do ? I do this : condition field=destination_number expression=^1#(d+)#(d+)$/ but it's not work ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] can't pass full sip url to dialplan
Hi:
I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:
*X-lite(softphone) dials - 1...@4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with 1009 (without the
sip: in front and without the @4.2.2.2 after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.
I would like to know how do you actually pass a full sip url to the dialplan
to do the regex match. Because from the default.xml dialplan, it comes with
an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume
there must be a way of passing full sip url to the dialplan. Here is the
example dialplan expecting sofia to pass it a full sip url:
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number expression=^sip:(.*)$
action application=bridge data=sofia/${use_profile}/$1/
/condition
/extension
Thanks
--
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Re: [Freeswitch-users] can't pass full sip url to dialplan
It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number expression=^sip:(.*)$
action application=bridge data=sofia/${use_profile}/$1/
/condition
/extension
And thanks for the tip, I will use variable instead.
On Sat, Aug 22, 2009 at 4:35 PM, Michael Jerris m...@jerris.com wrote:
No, you don't get the full sip uri in the dialplan like that. You do have
a whole bunch of variables of the parsed sip header you can use. Use the
info application to see all the vars so you can see what you have to route
the call on.
Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
Hi:
I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:
*X-lite(softphone) dials - 1...@4.2.2.2 (it's fake sip address, the
purpose was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with 1009 (without the
sip: in front and without the @4.2.2.2 after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.
I would like to know how do you actually pass a full sip url to the
dialplan to do the regex match. Because from the default.xml dialplan, it
comes with an example sip url dialing extension that match's *^sip:(.*)$ *.
So I assume there must be a way of passing full sip url to the dialplan.
Here is the example dialplan expecting sofia to pass it a full sip url:
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number expression=^sip:(.*)$
action application=bridge data=sofia/${use_profile}/$1/
/condition
/extension
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Re: [Freeswitch-users] can't pass full sip url to dialplan
Jason: I fully understand how the regex works in the dialplan. If you look closely in my original email and check out the pastebin. You will see that sofia does not pass the sip: to dialplan. I can do any combination of letters that dials from my softphone, and it will pass them to the dialplan. but if I put sip: in the front of my dial string. The sip: gets trunkated by sofia module so does the @xx.xx.xx.xx gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate sip: . Here is the excert from pastebin: 3.INVITE sip:1...@4.2.2.2 SIP/2.0 (line 3 , the freeswitch has successfuly received my dialying to sip: 1...@4.2.2.2)73. 2009-08-20 16:37:28.982772 [INFO] mod_dialplan_xml.c:315Processing 1001-1009 in context Global 74. Dialplan: sofia/trunkgroup_1/1...@192.168.1.67 parsing [ Global-number_1] continue=false 75. Dialplan: sofia/trunkgroup_1/1...@192.168.1.67 Regex (FAIL) [number_1]destination_number (1009) =~ /^sip(.*)$/ break=on-false (line 73~75, you can see that on line 73, sofia has trunkated the sip: @4.2.2.2 and only leave 1009 as the destination to pass to dialplan for regex match.) On Sat, Aug 22, 2009 at 6:30 PM, Jason White ja...@jasonjgw.net wrote: Henry Huang red.rain.se...@gmail.com wrote: It that case, the example of dialing sip_uri in the dialplan/default.xml should be removed to prevent confusion. Because according to what you said, one can never be able to hit this extension: It is entirely possible to reach this extension, but notice that the sip: prefix is removed before the rest of the URI is used in calling the bridge application. If you don't understand why this dial-plan entry works, go back and read about regular expressions and the format of destinations used with the bridge application. There are examples and explanations on the wiki. ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] can't pass full sip url to dialplan
Brian: but why can't I pass sip: to dialplan? seems like it's being truncated by sofia.. Can you confirm that? On Sat, Aug 22, 2009 at 10:30 PM, Brian West br...@freeswitch.org wrote: Remember the dialplan is agnostic... it has no clue about SIP, IAX, Jingle, H323... it routes... you have various other variables you can condition on also... route on destination_number and you'll be fine. /b On Aug 22, 2009, at 9:09 AM, Henry Huang wrote: I fully understand how the regex works in the dialplan. If you look closely in my original email and check out the pastebin. You will see that sofia does not pass the sip: to dialplan. I can do any combination of letters that dials from my softphone, and it will pass them to the dialplan. but if I put sip: in the front of my dial string. The sip: gets trunkated by sofia module so does the @xx.xx.xx.xx gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate sip: . ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] can't pass full sip url to dialplan
Brian: Sorry, it's my English. I didn't understand what you meant by agnostic back there. Now I know. Thank you. On Sat, Aug 22, 2009 at 10:59 PM, Brian West br...@freeswitch.org wrote: Because the dial plan is technology agnostic... you have been told more than once it won't pass it to the dialplan from mod_sofia... /b On Aug 22, 2009, at 9:46 AM, Henry Huang wrote: Brian: but why can't I pass sip: to dialplan? seems like it's being truncated by sofia.. Can you confirm that? ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] can't pass full sip url to dialplan
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample sip uri extension in the default.xml dialplan. because
no one can reach the dialplan with prefix sip: because sofia is going to
remove that prefix.
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number expression=^sip:(.*)$
action application=bridge data=sofia/${use_profile}/$1/
/condition
/extension
On Sat, Aug 22, 2009 at 10:59 PM, Brian West br...@freeswitch.org wrote:
Because the dial plan is technology agnostic... you have been told
more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass sip: to dialplan? seems like it's being
truncated by sofia..
Can you confirm that?
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Re: [Freeswitch-users] can't pass full sip url to dialplan
Michael:
Thank you for making it in for dummies format. :P
These are really nice tips I can use. thanks.
On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins m...@freeswitch.orgwrote:
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang red.rain.se...@gmail.comwrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample sip uri extension in the default.xml dialplan. because
no one can reach the dialplan with prefix sip: because sofia is going to
remove that prefix.
Well, this isn't entirely accurate. Like Mike J said, if you dialed
something like this at the CLI:
pa call sip:u...@domain.com sip%3au...@domain.com
Then you'd need the dialplan entry that handles the SIP URI.
Going back to the original question...
X-Lite dials 1...@4.2.2.2 correct?
But you're saying that the dialplan simply sees 1009 as the destination
number? I'm looking at the pastebin (10089) and trying to figure out exactly
what is happening. All I can see is that you have a context named Global
so I'm assuming you've made at least some modifications to the default
dialplan. Can you pastebin that whole context?
The other thing that you should probably do is create an extension in this
global context that routes a call to the info application. You could do
something like this so that 9992 would do an info dump:
extension name=info
condition field=destination_number expression=^(9992)$
action application=info/
/condition
/extension
Then reloadxml and make a call to 9992 from your X-Lite client. The CLI
will have a dump and you'll see all sorts of variables listed. Many of those
are available for you to use for condition matches and routing in the
dialplan.
Let us know how the info application does in giving you information about
the A leg of the call.
-MC
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number expression=^sip:(.*)$
action application=bridge data=sofia/${use_profile}/$1/
/condition
/extension
On Sat, Aug 22, 2009 at 10:59 PM, Brian West br...@freeswitch.orgwrote:
Because the dial plan is technology agnostic... you have been told
more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass sip: to dialplan? seems like it's being
truncated by sofia..
Can you confirm that?
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[Freeswitch-users] Dialing SIP URL issue
Hi:
I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:
*X-lite(softphone) dials - 1...@4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with 1009 (without the
sip: in front and without the @4.2.2.2 after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.
I would like to know how do you actually pass a full sip url to the dialplan
to do the regex match. Because from the default.xml dialplan, it comes with
an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume
there must be a way of passing full sip url to the dialplan. Here is the
example dialplan expecting sofia to pass it a full sip url:
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number expression=^sip:(.*)$
action application=bridge data=sofia/${use_profile}/$1/
/condition
/extension
Thanks
--
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[Freeswitch-users] Polycom Phone issue
I have setup 2 Freeswitch test server, 2 IP430 polycom phones with 2 lines
each registered to different server.
When I dial extension from the first line (first FS server), if the called
party hit reject on the phone. The called won't be hanged up will keep
ringing until timed out.
When dialed from second line (2nd FS server), the other phone would ring on
the screen as of the first line is ringing and besides the caller-id and
caller-id-name, it will also show the sip: address of the caller. Now it has
similar sympton to the above senario, but more over, even if leg B picks up
the call then hang up, Leg A (caller's end) will not hang up for as long as
you don't hang it up.
Do any of you people with polycom phone have problem like this ? or know
what could be the cause?
my FS1 server is version 12242M , FS2 server version is 13523M
*Here is my dialplan example of the extension that I called:*
extension name=eric
condition field=destination_number expression=^(2025)$/
condition field=caller_id_number expression=^(\d{2})(\d{2})$
action application=set data=caller_id_number=some_number/
action application=set data=call_timeout=35/
action application=bridge data=user/2025@
${domain_name},openzap/1/a/some_number/
anti-action application=set data=caller_id_number=some_number/
anti-action application=set data=call_timeout=35/
anti-action application=bridge data=user/2025@
${domain_name},openzap/1/a/some_number/
/condition
/extension
*
And here is the setting for register 2 server on polycom phone:*
reg.1.displayName=2025
reg.1.address=2025
reg.1.label=2025
reg.1.type=private
reg.1.lcs=
reg.1.thirdPartyName=
reg.1.auth.userId=2025
reg.1.auth.password=somepassword
reg.1.server.1.address=10.48.5.83
reg.1.server.1.port=5060
reg.1.server.1.transport=DNSnaptr
reg.1.server.2.transport=DNSnaptr
reg.1.server.1.expires=
reg.1.server.1.expires.overlap=
reg.1.server.1.register=1
reg.1.server.1.retryTimeOut=
reg.1.server.1.retryMaxCount=
reg.1.server.1.expires.lineSeize=
reg.1.outboundProxy.address=
reg.1.outboundProxy.port=
reg.1.outboundProxy.transport=
reg.1.acd-login-logout=0
reg.1.acd-agent-available=0
reg.1.proxyRequire=
reg.1.ringType=12
reg.1.lineKeys=1
reg.1.callsPerLineKey=
reg.2.displayName=2025
reg.2.address=2025
reg.2.label=2025
reg.2.type=private
reg.2.lcs=
reg.2.thirdPartyName=
reg.2.auth.userId=2025
reg.2.auth.password=somepassword
reg.2.server.1.address=10.55.7.36
reg.2.server.1.port=5060
reg.2.server.1.transport=DNSnaptr
reg.2.server.2.transport=DNSnaptr
reg.2.server.1.expires=
reg.2.server.1.expires.overlap=
reg.2.server.1.register=
reg.2.server.1.retryTimeOut=
reg.2.server.1.retryMaxCount=
reg.2.server.1.expires.lineSeize=
reg.2.outboundProxy.address=
reg.2.outboundProxy.port=
reg.2.outboundProxy.transport=
reg.2.acd-login-logout=0
reg.2.acd-agent-available=0
--
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VoIP Open Source software Consultant
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Re: [Freeswitch-users] Testimonials
Deigo: Do you work on some GUI interface or purely text environment? On Tue, May 19, 2009 at 4:19 PM, Diego Viola diego.vi...@gmail.com wrote: I'm also looking to install more PBXs with it soon, because FreeSWITCH does a great job at that, and at everything else too :) Diego On Tue, May 19, 2009 at 7:18 PM, Diego Viola diego.vi...@gmail.com wrote: Hi Maxim, We are using FreeSWITCH as an office PBX in Rich Apps Consulting (http://www.richappsconsulting.com/). I have written a blog post entry a long time ago, it's pretty old by now, but I guess still relevant. http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ It works great, I also have commented on the testimonial article, in the wiki about it. http://wiki.freeswitch.org/wiki/Testimonials#Office_PBX Regards, Diego On Tue, May 19, 2009 at 3:49 AM, Maxim Tsvetov maxim.tsve...@gmail.com wrote: Hello, Our company want to use Freeswitch and now we testing this solution in the lab. If someone already using Freeswitch as office pbx, ivr or any other commercial purposes could you please share you experience (and if it is not a secret - company name)? Regards, Maxim Tsvetov -- View this message in context: http://www.nabble.com/Testimonials-tp23611131p23611131.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 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] FS failover redundancy load balancing
How do you load balance conference calls? Doesn't all the conference members have to be on the same freeswitch server? On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko so...@gcdf.pl wrote: Brian West pisze: what kind of hardware? I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those results, it was over 100 calls that was handle good, I was just curios what will happen. Tomorrow I will make real testes. My production works on 2 core P4 and I have there only 35 agents CPU load is like 7% with 15% small peeks. All phones are sip or analog via sip gateways, PRI is currently still on asterisk which is connected via sip. /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] Lunchbox-type PC as small server?
Check out Shuttle XPC, they have a room for video card and a PCI slot. But you have to think a about reliability when deployed in business environment. I am using this as my home server. On Fri, Mar 6, 2009 at 8:30 AM, Fred codecompl...@free.fr wrote: Hello I'm looking for a small, lunchbox-like PC to build a small-form factor CRM server to sell to small companies. Ideally, it should have one PCI slot so that I can stick a voice card to connect to an analog phone line and run FreeSwitch as well. I like Asus' EeeBox (www.asus.com/products.aspx?l1=24l2=165) but it doesn't have room for a PCI slot, and I'm concerned about its performance. I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit pricey, and might also not be fast enough to act as a server. Are there brands/models you think I should look at? Thank you. ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] Detecting the origin of voice activity using VAD
Does the freeswitch VAD is able to distinguish ring tone from human voice? The scenario is to originate a call to a IVR system(don't connect the other leg here yet) and dial DTMF to get to the designated extension number , once someone picks up and say hello ( detected by VAD) now release to connect the other leg of the call. The point is to hold the first leg till a real person picks up. If it can't be done by VAD, how should I approach this function that I want to achieve. Thanks On Sun, Mar 1, 2009 at 1:15 PM, Brian West br...@freeswitch.org wrote: If i'm not mistaken those events will have a member-id in them so you can tell who they belong to. /b On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote: Using voice activity detection (VAD) in FreeSWITCH, how might I then distinguish which side of a call any given TALK or NOTALK event relates to? I am interested not just that there is activity on the call, but interested also in which party on the call is speaking (or not). Cam ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] Detecting the origin of voice activity using VAD
Well, I knew it would be some future fantasy for now.. If not human detection. I guess will try to use Dialplan Tools wait for silence to wait till the ring tone is finished ,then connect the other leg. On Sun, Mar 1, 2009 at 6:28 PM, Brian West br...@freeswitch.org wrote: NO. You want something that people THINK exists and works well... Reliable human/voice detection doesn't exist in ANY form. /b On Mar 1, 2009, at 8:20 PM, Henry Huang wrote: Does the freeswitch VAD is able to distinguish ring tone from human voice? The scenario is to originate a call to a IVR system(don't connect the other leg here yet) and dial DTMF to get to the designated extension number , once someone picks up and say hello ( detected by VAD) now release to connect the other leg of the call. The point is to hold the first leg till a real person picks up. If it can't be done by VAD, how should I approach this function that I want to achieve. 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] Detecting the origin of voice activity using VAD
ignore_early_media=true is not going to do the trick since once the IVR picks up the call on leg A, the ring tone is stopped and the IVR is going to play pre-recorded voice menu. And the freeswtich is going to send DTMF to reach a certain extension number say 101. Then the ring tone is going to start again while the IVR is going to dial the 101 extension(or even play moh while dialing). After extension 101 picks up, this is when I want the originate to connect call leg B on some other number. On Sun, Mar 1, 2009 at 7:11 PM, Brian West br...@freeswitch.org wrote: Usually ringing is done in early media... so the best bet would be to ignore_early_media=true /b On Mar 1, 2009, at 9:05 PM, Henry Huang wrote: Well, I knew it would be some future fantasy for now.. If not human detection. I guess will try to use Dialplan Tools wait for silence to wait till the ring tone is finished ,then connect the other leg. ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] snd_dummy setting for skype
I went through the wiki on mod_skypiax and see there should be a script to
make skype work without sound card in linux. Does anyone know where to
obtain that script to make sound work without sound card?
I am currently creating a /etc/asound.conf for skype to load the fake
sound driver. I do hear sound, but it's not perfect, it's very choppy and it
gives me error message when starting skype. The following is my asound.conf
setting. Hopefully someone can shed some light :
pcm.plugfile{
type plug
slave {
pcm infile
format S16_LE
channels 1
rate 16000
}
}
pcm.infile {
type file
slave {
pcm null
}
file /dev/dsp
infile /dev/dsp
}
by using this configuration. skype spit out error messages as follow but
still works:
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL plugfile
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL infile
--
Henry Huang
UniC Solution - Communication Unified
VoIP Open Source software Consultant
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Re: [Freeswitch-users] Deployment information and use cases
bandwidth.com has a service called phonebooth which is developed upon freeswitch. On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso r...@etellicom.com wrote: Hello FreeSWITCHERS, My company is currently creating a suite of applications which uses FreeSWITCH as the back-end for an IP-PBX solution. We currently have a prospect to have our first customer installation - a governmental department. That is a tender to have an IP-PBX installation to connect their four office branches, each one with about 300 users - which I am sure FreeSWITCH is able to handle. Since this is an official tender, it's part of their protocol to ask about real sites using the product. Having said that, would you mind sharing some information about your experience with FreeSWITCH deployments ? No need to give many details, but a short summary with company name (if possible), when it was deployed, server equipment, number of users, number of concurrent calls, what kind of functions and services are used and overall capacity of the system. I would really appreciate if you can share that information. And if you guys agree (and explicitly manifest your agreement), I can compile the information in the FreeSWITCH wiki under a Use Cases page so it can serve as a common reference as well. Kind regards, Raul Fragoso ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH VPSs
I run /usr/local/freeswitch/bin/freeswitch but I don't see a place where it says it's 32bit or 64bit. at the end of the initial script, I do see a version statement though. FreeSWITCH Version 1.0.trunk (exported) Started. Is there other ways to check if it's 32bit or 64bit? On Wed, Feb 11, 2009 at 6:55 PM, Brian West br...@freeswitch.org wrote: ding ding ding .. yep! file /usr/local/freeswitch/bin/freeswitch will also confirm /b On Feb 11, 2009, at 6:37 PM, Henry Huang wrote: Brian: I am also running my freeswitch on my own openVZ containers. Just how do you verify if the freeswitch is compiled as 64bit? I would assume if I compile it under a 64bit container, I would automatically get a 64bit freeswitch right? ___ 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 -- Henry Huang UniC Solution - Communication Unified ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH VPSs
Thinak you, William and Brian I got it now, I didn't know file was a command before because it didn't come with my CentOS installation. Now I have installed the file package and able to see the file info. Thanks again On Thu, Feb 12, 2009 at 12:31 PM, Brian West br...@freeswitch.org wrote: Well when I do this: r...@taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk 13:file /usr/local/freeswitch/bin/freeswitch /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked (uses shared libs), for GNU/Linux 2.6.9, not stripped It should clearly tell you. run the file command on it. /b On Feb 12, 2009, at 2:13 PM, Henry Huang wrote: I run /usr/local/freeswitch/bin/freeswitch but I don't see a place where it says it's 32bit or 64bit. at the end of the initial script, I do see a version statement though. FreeSWITCH Version 1.0.trunk (exported) Started. Is there other ways to check if it's 32bit or 64bit? ___ 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 -- Henry Huang UniC Solution - Communication Unified ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH VPSs
Brian: I am also running my freeswitch on my own openVZ containers. Just how do you verify if the freeswitch is compiled as 64bit? I would assume if I compile it under a 64bit container, I would automatically get a 64bit freeswitch right? On Wed, Feb 11, 2009 at 2:34 PM, Nik Martin freeswi...@servercorps.comwrote: If any one needs a FreeSWITCH box with a public, static IP, I can provide them for you at a reasonable cost. I'm building a Virtualization platform for FreeSWITCH hosting, and have the first node complete. These are OpenVZ Virtual Engines with Centos 5.2, a full build environment, and the latest FreeSWITCH trunk. You get 1 static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of bandwidth. Great for VOIP service providers, backup switch, testing, etc. You can contact me directly if you are interested. Nik ___ 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 -- Henry Huang UniC Solution - Communication Unified ___ 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
