Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?
Thanks for the link to PORTech's MV-370. For those interested, it can be had for a retail price of £150/€165 (before VAT). www.discountphonesystems.co.uk/acatalog/MV-370.html -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530601.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] Affordable GSM gateway for one cellphone?
What about this one? http://www.gempro.com.tw/gp-710.htm On Tue, Sep 22, 2009 at 11:07 AM, Fred-145 codecompl...@free.fr wrote: Thanks for the link to PORTech's MV-370. For those interested, it can be had for a retail price of £150/€165 (before VAT). www.discountphonesystems.co.uk/acatalog/MV-370.html -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530601.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 - Dmitry Bely ___ 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] Affordable GSM gateway for one cellphone?
I used Portech now for 3 years and have been very happy with their product. -E ___ 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] Mod_perl $session in not hangup
Hi all, I've the following mod_perl program to execute when I call to an
extension (say 777).
I use twinkle as a soft phone, to make calls.
#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
$session-answer();
if($session-ready())
{
my $uuid=$session-getVariable(uuid);
freeswitch::consoleLog(INFO,UUID is $uuid\n);
freeswitch::consoleLog(INFO,Session is answered\n);
$session-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav);
my $dtmf = $session-getDigits(4,, 5000);
freeswitch::consoleLog(INFO,I received $dtmf\n);
$session-hangup(NORMAL_CLEARING);
sleep(5);
# Some other statements.
}
return 1;
Everything is fine.
After executing $session-hangup, I got NORMAL_CLEARING in my freeswitch
console. But in my soft phone, still the channel is active for 5 seconds.
The call got ended only after the 5 seconds sleep.
But if I create my own session like
my $session=new
freeswitch::Session(user/1000);
and I say $session-hangup(), it got terminated.
I wanted to know why there is such difference?? or am I wrong??
Please clarify me.
--
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http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646.html
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Re: [Freeswitch-users] Displaying matched extension during a call
Thanks Michael for your links. Till now I was working on the command line
which we get on executing the 'freeswitch' command. I didn't know abt the
'fs_cli' command. The fs_cli gives lot more information that will be helpful
for novice user like me.
Thanks,
Anil
2009/9/22 Michael Collins m...@freeswitch.org
On Mon, Sep 21, 2009 at 11:27 AM, Anil Kumar S. R. sra...@gmail.comwrote:
Hi All,
I am new to Freeswitch. So please bear with me if I ask any silly
questions.
* Can anyone of you please tell me how to display the extension name which
has matched an incoming/outgoing call.
* And can you please elaborate what does 'action application=info/'
mean.
* Suppose we have set a variable in the extension of the dialplan XML. Is
there anyway we can display this variable on CLI for our debugging purposes.
Anil,
Here are few links to get started:
Handy tutorial: http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
Chan vars: http://wiki.freeswitch.org/wiki/Channel_Variables
Log app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log
Note: the info app dumps all sorts of information to the console and is a
great way to learn about many of the channel variables that FS has. The log
app will make it easy for you to pinpoint just a single channel variable:
action application=log data=INFO Dialed extension is ${dialed_ext}/
Have fun!
-MC
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The best way to succeed in this world is to act on the advice you give to
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Re: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described
below, but I still get the same timeout error:
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed
Registration, setting retry to 270 seconds.
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration
Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
include
gateway name=asterisk
param name=username value=28/
param name=realm value=192.168.1.119/
param name=proxy value=192.168.1.119/
param name=password value=test/
param name=register value=true/
param name=caller-id-in-from value=true/
param name=sip-port value=5060/param
/gateway
/include
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param name=realm value=ASTERISK_IP_ADDRESS/
param name=password value=PASSWORD_ON_ASTERISK/
param name=register value=true/
param name=caller-id-in-from value=true/
/gateway
/include
this should be enough to register the GW... after that please read
this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
in your case it will be something like this:
extension name=dialGW
condition field=destination_number
expression=^(NUMBER_TO_SEND_TO_ASTERISK)$
action application=bridge data=sofia/gateway/gw01/$1/
/condition
/extension
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows
installation.
Now Im trying to get my 2 Sipphones get connected to. Later I want
connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a
simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each
other (
registration seems to work now).
the next is , that freeshwitch is not able to connect to asterisk.
but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the
following content :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number
value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number
value=24/variable
/variables
/user
/domain
/include
This seems to be ok now. Now I want to dial from 22 to 24 ,
wherefore I
configured this dialplan :
include
context name=any
condition field=destination_number expression=^(2[0-9])$
action application=bridge data=user/${dialed_extension}/
/condition
/include
wich doesnt work , mybe b/c the user/${dialed_extension} I dont
know...
Freeswitch says:
[INFO] switch_core_state_machine.c:136 No Route, Aborting
[NOTICE] switch_core_state_machine.c:137 Hangup
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34
[CS_ROUTING] [NO_ROUTE_DESTINATION]
[NOTICE] switch_core_session.c:1086 Session 17
(sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34) Ended
[NOTICE] switch_core_session.c:1088 Close Channel
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34 [CS_DESTROY]
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help me out here
to .. :
In the sip-profiles/external I installed the my_asterisk.xml with that
content :
include
gateway name=asterisk
param name=username value=28/param
param name=password value=test/param
param name=realm value=28/param
param name=proxy value=192.168.1.119/param
Re: [Freeswitch-users] Mod_perl or ESL
I need to handle some hundreds of call. So, which one can I opt? Nagalenoj wrote: Dear friends, I want to know which is the better way to do route calls and control calls. I've did a experiment which can be done in both ways, Mod_perl and ESL. I don't know which one is better to take. When I see some earlier posts, It is given like Mod_perl has some limitations and I don't know what kind of limitations they are., Can someone say which is better to use and how it is better? -- View this message in context: http://www.nabble.com/Mod_perl-or-ESL-tp25520023p25530677.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] FreeSWITCH - UTRAN/UTRA
hi all, Has anyone connected FreeSWITCH to UTRAN? If so, how was that done? Jan _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx___ 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] Unable to set internal call to registered sip user
Dear List,
I read the documentation, but Im still confused about how to dial a
internal registered sip user.
I configured the both sip phones in the directory in my local.xml file :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number value=24/variable
/variables
/user
/domain
/include
It seems, that they can connect to the freeswitch.
I configured the dialplan like following :
include
context name=default
extension name=diallocal
condition field=destination_number expression=^(2[0-9])$
!--- The % behind the username tells FS to lookup the user in
it's local sip_registration database --
action application=bridge
data=user/${dialed_extensi...@${domain_name}/action
!--- x.x.x.x in the line above is the IP address to the
FreeSWITCH server/device --
!--- If you don't want to bridge a call to a local registered
user, but to a SIP URI, use the @ instead of %:
action application=bridge
data=sofia/profilename/5...@x.x.x.x/ --
/condition
/extension
...
If I call from the sip user 24 to 22 , freeswitch logs the following and
gives an busy tone immediately:
freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
[decc119c-a973-6b4c-bf11-ec251c653cda]
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing
24-22 in context default
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user
[...@192.168.1.34]
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: SUBSCRIBER_ABSENT
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup
sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session
13 (sofia/internal/2...@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY]
thanks again for your help ...
regards,
Filip
--
_
Filip Lyncker, Dipl.-Inform. (FH)
Lyncker Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany
Fon +49 611/9006951
Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399
Geschäftsführer:
Filip Lyncker,
Armin Theis
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Re: [Freeswitch-users] Unable to set internal call to registered sip user
and this is not enough for you?
!--- The *%* behind the username tells FS to lookup the user in it's
local sip_registration database --
action application=bridge data=user/${dialed_extension}@
${domain_name}/
!--- x.x.x.x in the line above is the IP address to the FreeSWITCH
server/device --
!--- If you don't want to bridge a call to a local registered user,
but to a SIP URI, use the @ instead of %:
action application=bridge data=sofia/profilename/5...@x.x.x.x/
--
T.
On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@lyth.de wrote:
Dear List,
I read the documentation, but Im still confused about how to dial a
internal registered sip user.
I configured the both sip phones in the directory in my local.xml file :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number value=24/variable
/variables
/user
/domain
/include
It seems, that they can connect to the freeswitch.
I configured the dialplan like following :
include
context name=default
extension name=diallocal
condition field=destination_number expression=^(2[0-9])$
!--- The % behind the username tells FS to lookup the user in
it's local sip_registration database --
action application=bridge
data=user/${dialed_extensi...@${domain_name}/action
!--- x.x.x.x in the line above is the IP address to the
FreeSWITCH server/device --
!--- If you don't want to bridge a call to a local registered
user, but to a SIP URI, use the @ instead of %:
action application=bridge
data=sofia/profilename/5...@x.x.x.x/ --
/condition
/extension
...
If I call from the sip user 24 to 22 , freeswitch logs the following and
gives an busy tone immediately:
freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
[decc119c-a973-6b4c-bf11-ec251c653cda]
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing
24-22 in context default
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user
[...@192.168.1.34]
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: SUBSCRIBER_ABSENT
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup
sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session
13 (sofia/internal/2...@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY]
thanks again for your help ...
regards,
Filip
--
_
Filip Lyncker, Dipl.-Inform. (FH)
Lyncker Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany
Fon +49 611/9006951
Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399
Geschäftsführer:
Filip Lyncker,
Armin Theis
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Re: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration
hmmm .. can you register using x-lite or some other softphone with the same
credentials?
can you paste a siptrace of the failed registration?
BTW: Make sure nothing is already registered with this credentials when you
try with FS
T.
On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker lync...@lyth.de wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described
below, but I still get the same timeout error:
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed
Registration, setting retry to 270 seconds.
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration
Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
include
gateway name=asterisk
param name=username value=28/
param name=realm value=192.168.1.119/
param name=proxy value=192.168.1.119/
param name=password value=test/
param name=register value=true/
param name=caller-id-in-from value=true/
param name=sip-port value=5060/param
/gateway
/include
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param name=realm value=ASTERISK_IP_ADDRESS/
param name=password value=PASSWORD_ON_ASTERISK/
param name=register value=true/
param name=caller-id-in-from value=true/
/gateway
/include
this should be enough to register the GW... after that please read
this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
in your case it will be something like this:
extension name=dialGW
condition field=destination_number
expression=^(NUMBER_TO_SEND_TO_ASTERISK)$
action application=bridge data=sofia/gateway/gw01/$1/
/condition
/extension
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows
installation.
Now Im trying to get my 2 Sipphones get connected to. Later I want
connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a
simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each
other (
registration seems to work now).
the next is , that freeshwitch is not able to connect to asterisk.
but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the
following content :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number
value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number
value=24/variable
/variables
/user
/domain
/include
This seems to be ok now. Now I want to dial from 22 to 24 ,
wherefore I
configured this dialplan :
include
context name=any
condition field=destination_number expression=^(2[0-9])$
action application=bridge data=user/${dialed_extension}/
/condition
/include
wich doesnt work , mybe b/c the user/${dialed_extension} I dont
know...
Freeswitch says:
[INFO] switch_core_state_machine.c:136 No Route, Aborting
[NOTICE] switch_core_state_machine.c:137 Hangup
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34
[CS_ROUTING] [NO_ROUTE_DESTINATION]
[NOTICE] switch_core_session.c:1086 Session 17
(sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34) Ended
[NOTICE] switch_core_session.c:1088 Close Channel
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34 [CS_DESTROY]
Im sure , for you
Re: [Freeswitch-users] recompile with gdb
see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga tculj...@gmail.com wrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West br...@freeswitch.org wrote: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 node1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# Of course I'm using the latest trunk... Can anyone help? ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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] Can this be done in FreeSWITCH?
Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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] Unable to set internal call to registered sip user
ok , i tried several things :
action application=bridge
data=user/${dialed_extension}%${domain_name}/action
action application=bridge
data=user/${dialed_extension}%192.168.1.34/action
action application=bridge data=user/${dialed_extension}/action
but all this doesnt work sorry mybe I dont see something apparent , but
I dont have a clue...
Tihomir Culjaga schrieb:
and this is not enough for you?
!--- The *%* behind the username tells FS to lookup the user in
it's local sip_registration database --
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
!--- x.x.x.x in the line above is the IP address to the
FreeSWITCH server/device --
!--- If you don't want to bridge a call to a local registered
user, but to a SIP URI, use the @ instead of %:
action application=bridge
data=sofia/profilename/5...@x.x.x.x/ --
T.
On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de wrote:
Dear List,
I read the documentation, but Im still confused about how to dial a
internal registered sip user.
I configured the both sip phones in the directory in my local.xml
file :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number
value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number
value=24/variable
/variables
/user
/domain
/include
It seems, that they can connect to the freeswitch.
I configured the dialplan like following :
include
context name=default
extension name=diallocal
condition field=destination_number expression=^(2[0-9])$
!--- The % behind the username tells FS to lookup the user in
it's local sip_registration database --
action application=bridge
data=user/${dialed_extensi...@${domain_name}/action
!--- x.x.x.x in the line above is the IP address to the
FreeSWITCH server/device --
!--- If you don't want to bridge a call to a local registered
user, but to a SIP URI, use the @ instead of %:
action application=bridge
data=sofia/profilename/5...@x.x.x.x/ --
/condition
/extension
...
If I call from the sip user 24 to 22 , freeswitch logs the
following and
gives an busy tone immediately:
freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
mailto:2...@192.168.1.34
[decc119c-a973-6b4c-bf11-ec251c653cda]
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing
24-22 in context default
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find
user
[...@192.168.1.34 http://192.168.1.34]
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: SUBSCRIBER_ABSENT
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34
[CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session
13 (sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34
[CS_DESTROY]
thanks again for your help ...
regards,
Filip
--
_
Filip Lyncker, Dipl.-Inform. (FH)
Lyncker Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany
Fon +49 611/9006951
Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399
Geschäftsführer:
Filip Lyncker,
Armin Theis
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Re: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration
Ok *solved* I set in my sip.conf (asterisk) now nat=true, b/c the
asterisk ansered the packets sent from lan_ip to the external_ip.
now it works, but its not the perfect solution because FS seems to send
the packets with an nat envelope or flag. How can i avoid this?
the next thing is the dialplan, wich doesnt work at all for me ! ( see
my other post with sip registrares) ... if I call now a number , the
following entry should route it to my asterisk-gw :
context name=any
extension name=dialasterisk
condition field=destination_number expression=^${dialed_extension}$
action application=bridge data=sofia/gateway/asterisk/$1/
/condition
/extension
/context
but it doesnt and FS says :
freeswi...@bigfish 2009-09-22 17:10:16.776629 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
[733236b0-be36-0049-8ace-a2903921fd81]
2009-09-22 17:10:16.781511 [INFO] mod_dialplan_xml.c:315 Processing
22-01776721280 in context default
2009-09-22 17:10:16.800065 [NOTICE] switch_ivr.c:1349 Transfer
sofia/internal/2...@192.168.1.34 to enum[01776721...@default]
2009-09-22 17:10:26.800401 [INFO] switch_core_state_machine.c:136 No
Route, Aborting
2009-09-22 17:10:26.800401 [NOTICE] switch_core_state_machine.c:137
Hangup sofia/internal/2...@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2009-09-22 17:10:26.800401 [NOTICE] switch_core_session.c:1086 Session 3
(sofia/internal/2...@192.168.1.34) Ended
2009-09-22 17:10:26.800401 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY]
what's wrong with my dialplan ?
thanks again for help,
regards
filip
Tihomir Culjaga schrieb:
hmmm .. can you register using x-lite or some other softphone with the
same credentials?
can you paste a siptrace of the failed registration?
BTW: Make sure nothing is already registered with this credentials
when you try with FS
T.
On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described
below, but I still get the same timeout error:
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed
Registration, setting retry to 270 seconds.
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk
Registration
Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
include
gateway name=asterisk
param name=username value=28/
param name=realm value=192.168.1.119/
param name=proxy value=192.168.1.119/
param name=password value=test/
param name=register value=true/
param name=caller-id-in-from value=true/
param name=sip-port value=5060/param
/gateway
/include
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param name=realm value=ASTERISK_IP_ADDRESS/
param name=password value=PASSWORD_ON_ASTERISK/
param name=register value=true/
param name=caller-id-in-from value=true/
/gateway
/include
this should be enough to register the GW... after that please read
this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
in your case it will be something like this:
extension name=dialGW
condition field=destination_number
expression=^(NUMBER_TO_SEND_TO_ASTERISK)$
action application=bridge data=sofia/gateway/gw01/$1/
/condition
/extension
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de
mailto:lync...@lyth.de mailto:lync...@lyth.de wrote:
Hi List,
for the first experiments with freeswitch I downloaded the
Windows
installation.
Now Im trying to get my 2 Sipphones get connected to. Later
I want
connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to
build up a
simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each
other (
registration seems to work now).
the next is , that freeshwitch is not able to connect to
asterisk.
but I
will describe this later.
I installed in the Directory a xml file ( called
Re: [Freeswitch-users] Mod_perl $session in not hangup
The reason is you cannot complete the hangup until the script exits.
On the bright side, if you update to latest trunk it will probably work more
how you want it to
because a recent change will make this possible.
On Tue, Sep 22, 2009 at 4:30 AM, lakshmanan lakindi...@gmail.com wrote:
Hi all, I've the following mod_perl program to execute when I call to an
extension (say 777).
I use twinkle as a soft phone, to make calls.
#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
$session-answer();
if($session-ready())
{
my $uuid=$session-getVariable(uuid);
freeswitch::consoleLog(INFO,UUID is $uuid\n);
freeswitch::consoleLog(INFO,Session is answered\n);
$session-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav);
my $dtmf = $session-getDigits(4,, 5000);
freeswitch::consoleLog(INFO,I received $dtmf\n);
$session-hangup(NORMAL_CLEARING);
sleep(5);
# Some other statements.
}
return 1;
Everything is fine.
After executing $session-hangup, I got NORMAL_CLEARING in my freeswitch
console. But in my soft phone, still the channel is active for 5 seconds.
The call got ended only after the 5 seconds sleep.
But if I create my own session like
my $session=new
freeswitch::Session(user/1000);
and I say $session-hangup(), it got terminated.
I wanted to know why there is such difference?? or am I wrong??
Please clarify me.
--
View this message in context:
http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
___
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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/
Twitter: http://twitter.com/FreeSWITCH_wire
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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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?
portech also seems to be good. On Sun, Sep 20, 2009 at 5:57 PM, Fred-145 codecompl...@free.fr wrote: Thanks Tihomir for the link. From what I read, it appears that EdgePBX's FX02G is a full-fledged Asterisk server with a GSM module and an FXS module. Did you reflash its NAND to run Freeswitch? At $300, I guess customers will rather take a subscription with a VoIP provided and use their GSM gateway, but I'm interested in knowing whether the FX02G can be used as a PSTN/GSM gateway, possibly with FreeSwitch running on that unit as well. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530130.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 -- Itamar Reis Peixoto e-mail/msn: ita...@ispbrasil.com.br sip: ita...@ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 ___ 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] Some Newbie questions about dialplan and local Sip registration
now i registered from my x-lite client without anyproblems.
but I think i got it now, my tcpdump says the following :
IP 192.168.1.119.5060 93.210.212.xxx.5080: SIP, length: 465
wich is the external IP of my network ! must have somthing todo with NAT
/ Masquerade options... how can I avoid this ?
thanks for your help ...
regards,
filip
Tihomir Culjaga schrieb:
hmmm .. can you register using x-lite or some other softphone with the
same credentials?
can you paste a siptrace of the failed registration?
BTW: Make sure nothing is already registered with this credentials
when you try with FS
T.
On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described
below, but I still get the same timeout error:
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed
Registration, setting retry to 270 seconds.
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk
Registration
Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
include
gateway name=asterisk
param name=username value=28/
param name=realm value=192.168.1.119/
param name=proxy value=192.168.1.119/
param name=password value=test/
param name=register value=true/
param name=caller-id-in-from value=true/
param name=sip-port value=5060/param
/gateway
/include
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param name=realm value=ASTERISK_IP_ADDRESS/
param name=password value=PASSWORD_ON_ASTERISK/
param name=register value=true/
param name=caller-id-in-from value=true/
/gateway
/include
this should be enough to register the GW... after that please read
this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
in your case it will be something like this:
extension name=dialGW
condition field=destination_number
expression=^(NUMBER_TO_SEND_TO_ASTERISK)$
action application=bridge data=sofia/gateway/gw01/$1/
/condition
/extension
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de
mailto:lync...@lyth.de mailto:lync...@lyth.de wrote:
Hi List,
for the first experiments with freeswitch I downloaded the
Windows
installation.
Now Im trying to get my 2 Sipphones get connected to. Later
I want
connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to
build up a
simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each
other (
registration seems to work now).
the next is , that freeshwitch is not able to connect to
asterisk.
but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml)
with the
following content :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number
value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number
value=24/variable
/variables
Re: [Freeswitch-users] recompile with gdb
FreeSWITCH compiles with debug symbols by default but you showed an output where GCC was segfaulting so you have bad ram or bad hardware and I suspect that is your problem. /b On Sep 22, 2009, at 7:36 AM, Tihomir Culjaga wrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mod_perl or ESL
either one will work. The drawback of mod_perl is that the code executes inline so run the risk of a mistake in your perl code making FreeSWITCH become less stable. The drawback of ESL is you are opening a socket connection for each call. On Tue, Sep 22, 2009 at 6:11 AM, Nagalenoj nagale...@gmail.com wrote: I need to handle some hundreds of call. So, which one can I opt? Nagalenoj wrote: Dear friends, I want to know which is the better way to do route calls and control calls. I've did a experiment which can be done in both ways, Mod_perl and ESL. I don't know which one is better to take. When I see some earlier posts, It is given like Mod_perl has some limitations and I don't know what kind of limitations they are., Can someone say which is better to use and how it is better? -- View this message in context: http://www.nabble.com/Mod_perl-or-ESL-tp25520023p25530677.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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] Unable to set internal call to registered sip user
Hi
in dialplan i see:
condition field=destination_number expression=^(2[0-9])$ - check
on variable destination_number
and later
action application=bridge
data=user/${dialed_extensi...@${domain_name}/action -
bridge to variable dialed_extension , other then checked
destination_number
or $1 from regexp
try:
action application=bridge
data=user/${destination_number}%${sip_profile}/action
or
action application=bridge
data=user/$1%${sip_profile}/action
By
Kleo
On Tue, 22 Sep 2009, Filip Lyncker wrote:
Dear List,
I read the documentation, but Im still confused about how to dial a
internal registered sip user.
I configured the both sip phones in the directory in my local.xml file :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number value=24/variable
/variables
/user
/domain
/include
It seems, that they can connect to the freeswitch.
I configured the dialplan like following :
include
context name=default
extension name=diallocal
condition field=destination_number expression=^(2[0-9])$
!--- The % behind the username tells FS to lookup the user in
it's local sip_registration database --
action application=bridge
data=user/${dialed_extensi...@${domain_name}/action
!--- x.x.x.x in the line above is the IP address to the
FreeSWITCH server/device --
!--- If you don't want to bridge a call to a local registered
user, but to a SIP URI, use the @ instead of %:
action application=bridge
data=sofia/profilename/5...@x.x.x.x/ --
/condition
/extension
...
If I call from the sip user 24 to 22 , freeswitch logs the following and
gives an busy tone immediately:
freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
[decc119c-a973-6b4c-bf11-ec251c653cda]
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing
24-22 in context default
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user
[...@192.168.1.34]
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: SUBSCRIBER_ABSENT
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup
sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session
13 (sofia/internal/2...@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY]
thanks again for your help ...
regards,
Filip
--
_
| You have moved the mouse. #
| Windows must be restarted for the changes to take effect. #
| OK #
##/
~~ ~~ ~~ ~~ ~~ ~~ ~~
Vladimir `KLEO' Klejch Kleo'at'netbox.cz
... ... ... ...
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Re: [Freeswitch-users] ALLOTTED_TIMEOUT hangup cause?
Did a little more digging, ALLOTTED_TIMEOUT has an error code of 602 according to the Wiki (http://wiki.freeswitch.org/wiki/Hangup_causes) nevertheless that code is not covered in RFC 4497 ( http://tools.ietf.org/html/rfc4497) On Mon, Sep 21, 2009 at 8:41 PM, Nicolas Brenner nico...@medularis.comwrote: Hi, Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the Wiki and Google, but I couldn't find a detailed explanation. Does anybody know what does it mean exactly? Thanks! Nicolas ___ 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 this be done in FreeSWITCH?
How is the DID transported? SIP, PRI, analog DID trunks? -MC On Mon, Sep 21, 2009 at 9:17 PM, Francis Vidal francisv.l...@gmail.comwrote: Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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] Can this be done in FreeSWITCH?
well .. it is AS .. it can be SIP or H323 ... well if it is hooked to a PGW it is MGCP but i doubt... so it is either SIP or H323. i will put a nickel for H323 :P T. On Tue, Sep 22, 2009 at 6:49 PM, Tihomir Culjaga tculj...@gmail.com wrote: so, you say ... CallingParty = AS5300 A: aNum B: didNum AS5300 = PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. T. On Tue, Sep 22, 2009 at 6:17 AM, Francis Vidal francisv.l...@gmail.comwrote: Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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] Can this be done in FreeSWITCH?
so, you say ... CallingParty = AS5300 A: aNum B: didNum AS5300 = PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. T. On Tue, Sep 22, 2009 at 6:17 AM, Francis Vidal francisv.l...@gmail.comwrote: Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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] recompile with gdb
Why don't you try to do the same on another machine to see if you get the same results? I think it's hardware related as Anthony and Brian pointed out. Diego On Tue, Sep 22, 2009 at 4:29 PM, Tihomir Culjaga tculj...@gmail.com wrote: Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module So, to summarize.. i had two issues: 1. FS crashing without any notice (at 5 CPS) 2. Unable to recompile FS with gdb support The first issue was actually related to -hp switch i was using in my startup script. With it, FS was crashing without any notice (even on low traffic) and regardless if i load my custom modules or not. The second issue was related to many FS crashes having my module loaded... I found it later and fixed that. So, after the machine cleanup I rebuild FS with gdb support without any issues. Of course i sow this log .. but i didn't realize for a while... and after that i was fighting with crashes caused by -hp ... also, it was quite late as well ended up at 3 AM :P Anyhow, the poit is; FS works well with my custom module. It just finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... well, thats something :P. T. On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale anthony.miness...@gmail.com wrote: see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/ pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga tculj...@gmail.comwrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West br...@freeswitch.orgwrote: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 node1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# Of course I'm using the latest trunk... Can anyone help? ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] recompile with gdb
Why don't you try changing RAM? Or run memtest86 or try another machine? Or... On Tue, Sep 22, 2009 at 5:46 PM, Tihomir Culjaga tculj...@gmail.com wrote: hmmm, how to track that down? this is gonna be tricky... i have another machine but quite different i can try on that as well and we will see T. On Tue, Sep 22, 2009 at 6:41 PM, Brian West br...@freeswitch.org wrote: The issue is you clearly show GCC crashing trying to compile freeswitch which is BAD that indicates a larger problem with the hardware or memory. Its physical issues not logical ones. /b On Sep 22, 2009, at 11:29 AM, Tihomir Culjaga wrote: Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module ___ 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] recompile with gdb
One of the things that -hp does is call memlockall which disables swapping which uses more memory which makes hitting a land mine in your ram chip much more likely. On the other hand: Since you are talking about with and without gcc support I am going to guess you are on Solaris which you probably should have mentioned before. it's possible that some of the more aggressive things activated by -hp is not possible on that platform. If so we either have to identify that and disable it or disable hp completely for Solaris. Either way, gcc randomly crashing is never ok and is a symptom of a pretty serious issue. Are you using 2 separate fresh checkouts for both suncc and gcc builds because it's not possible to switch the same source tree once it's already configured for one of them. On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga tculj...@gmail.comwrote: Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module So, to summarize.. i had two issues: 1. FS crashing without any notice (at 5 CPS) 2. Unable to recompile FS with gdb support The first issue was actually related to -hp switch i was using in my startup script. With it, FS was crashing without any notice (even on low traffic) and regardless if i load my custom modules or not. The second issue was related to many FS crashes having my module loaded... I found it later and fixed that. So, after the machine cleanup I rebuild FS with gdb support without any issues. Of course i sow this log .. but i didn't realize for a while... and after that i was fighting with crashes caused by -hp ... also, it was quite late as well ended up at 3 AM :P Anyhow, the poit is; FS works well with my custom module. It just finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... well, thats something :P. T. On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale anthony.miness...@gmail.com wrote: see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/ pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga tculj...@gmail.comwrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West br...@freeswitch.orgwrote: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 node1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# Of course I'm using the latest trunk... Can anyone help? ___ 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
Re: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released
Nice link, Are you offering to become a reseller? On Mon, Sep 21, 2009 at 6:55 PM, Hadley Rich h...@nice.net.nz wrote: On Tue, 22 Sep 2009 11:13:13 Gavin Henry wrote: URL??? On 21/09/2009, Michael Collins m...@freeswitch.org wrote: We invite you to visit the CudaTel http://www.cudatel.com/ website or call 989-720-4000 for more information or to request evaluation units. -- https://nicegear.co.nz VoIP and Open Source Hardware ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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] CudaTel Communications Server Version 1.0 Released
On Wed, 23 Sep 2009 06:36:17 Anthony Minessale wrote: Nice link, Are you offering to become a reseller? Heh, I was actually trying to quote the original link to the person that asked for it rather than spam with my sig. That said, we'd love to be a reseller for our little part of the world. hads -- https://nicegear.co.nz VoIP and Open Source Hardware ___ 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] recompile with gdb
Hi, Nope, I'm still on Debian 5.0... in transit to CentOS 5.3 but it needs to wait a bit. i was talking about gdb, not gcc and was trying to recompile FS with debug symbols on: CFLAGS=-g -ggdb MOD_CFLAGS=-g -ggdb. yes, I understand that gcc segfault most probably means only one thing... HW isues. This is sometihng that I'm going to check tomorrow running memtest to see what i get. Also, I will repeat the same test with a new block of RAM. Maybe i didn't explain myself well... apologize. T. On Tue, Sep 22, 2009 at 8:42 PM, Anthony Minessale anthony.miness...@gmail.com wrote: One of the things that -hp does is call memlockall which disables swapping which uses more memory which makes hitting a land mine in your ram chip much more likely. On the other hand: Since you are talking about with and without gcc support I am going to guess you are on Solaris which you probably should have mentioned before. it's possible that some of the more aggressive things activated by -hp is not possible on that platform. If so we either have to identify that and disable it or disable hp completely for Solaris. Either way, gcc randomly crashing is never ok and is a symptom of a pretty serious issue. Are you using 2 separate fresh checkouts for both suncc and gcc builds because it's not possible to switch the same source tree once it's already configured for one of them. On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga tculj...@gmail.comwrote: Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module So, to summarize.. i had two issues: 1. FS crashing without any notice (at 5 CPS) 2. Unable to recompile FS with gdb support The first issue was actually related to -hp switch i was using in my startup script. With it, FS was crashing without any notice (even on low traffic) and regardless if i load my custom modules or not. The second issue was related to many FS crashes having my module loaded... I found it later and fixed that. So, after the machine cleanup I rebuild FS with gdb support without any issues. Of course i sow this log .. but i didn't realize for a while... and after that i was fighting with crashes caused by -hp ... also, it was quite late as well ended up at 3 AM :P Anyhow, the poit is; FS works well with my custom module. It just finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... well, thats something :P. T. On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale anthony.miness...@gmail.com wrote: see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/ pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga tculj...@gmail.comwrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West br...@freeswitch.orgwrote: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2
Re: [Freeswitch-users] Custom Variables
This should defiantly be in there, please double check if its in a different name, and if not, please post a bug to jira.freeswitch.org. Mike On Sep 8, 2009, at 5:27 PM, Tina Martinez wrote: Using the verbose-events definitely improved my ability to see the custom variables, but now I noticed that the Member-ID variable does not appear in the DTMF event. Would this be related, or did I screw something else up? - T ___ 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] SUBSCRIBE and NOTIFY
I'm having to configure FreeSWITCH. Baxei version 1.0.4 and I am accessing with the users 1000 and 1001. I register, make the connection. But I'm trying to see to see who is connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I connect the other User does not receive the information that I connected (I sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected to another user I'm not, sometimes you 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
Re: [Freeswitch-users] recompile with gdb
Doesn't FS already compiles with debug symbols by default? On Tue, Sep 22, 2009 at 7:04 PM, Tihomir Culjaga tculj...@gmail.com wrote: Hi, Nope, I'm still on Debian 5.0... in transit to CentOS 5.3 but it needs to wait a bit. i was talking about gdb, not gcc and was trying to recompile FS with debug symbols on: CFLAGS=-g -ggdb MOD_CFLAGS=-g -ggdb. yes, I understand that gcc segfault most probably means only one thing... HW isues. This is sometihng that I'm going to check tomorrow running memtest to see what i get. Also, I will repeat the same test with a new block of RAM. Maybe i didn't explain myself well... apologize. T. On Tue, Sep 22, 2009 at 8:42 PM, Anthony Minessale anthony.miness...@gmail.com wrote: One of the things that -hp does is call memlockall which disables swapping which uses more memory which makes hitting a land mine in your ram chip much more likely. On the other hand: Since you are talking about with and without gcc support I am going to guess you are on Solaris which you probably should have mentioned before. it's possible that some of the more aggressive things activated by -hp is not possible on that platform. If so we either have to identify that and disable it or disable hp completely for Solaris. Either way, gcc randomly crashing is never ok and is a symptom of a pretty serious issue. Are you using 2 separate fresh checkouts for both suncc and gcc builds because it's not possible to switch the same source tree once it's already configured for one of them. On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga tculj...@gmail.comwrote: Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module So, to summarize.. i had two issues: 1. FS crashing without any notice (at 5 CPS) 2. Unable to recompile FS with gdb support The first issue was actually related to -hp switch i was using in my startup script. With it, FS was crashing without any notice (even on low traffic) and regardless if i load my custom modules or not. The second issue was related to many FS crashes having my module loaded... I found it later and fixed that. So, after the machine cleanup I rebuild FS with gdb support without any issues. Of course i sow this log .. but i didn't realize for a while... and after that i was fighting with crashes caused by -hp ... also, it was quite late as well ended up at 3 AM :P Anyhow, the poit is; FS works well with my custom module. It just finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... well, thats something :P. T. On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale anthony.miness...@gmail.com wrote: see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/ pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga tculj...@gmail.comwrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West br...@freeswitch.orgwrote: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre'
Re: [Freeswitch-users] recompile with gdb
yes On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: Doesn't FS already compiles with debug symbols by default? ___ 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] recompile with gdb
Then why is Tihomir trying to compile with debug symbols? On Tue, Sep 22, 2009 at 8:00 PM, Brian West br...@freeswitch.org wrote: yes On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: Doesn't FS already compiles with debug symbols by default? ___ 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] recompile with gdb
He's doing an extra effort... just compile it as you would normally and you will have the debug symbols. On Tue, Sep 22, 2009 at 8:11 PM, Diego Viola diego.vi...@gmail.com wrote: Then why is Tihomir trying to compile with debug symbols? On Tue, Sep 22, 2009 at 8:00 PM, Brian West br...@freeswitch.org wrote: yes On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: Doesn't FS already compiles with debug symbols by default? ___ 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] recompile with gdb
well ... shame on me :P thx anyway... T. On Tue, Sep 22, 2009 at 10:12 PM, Diego Viola diego.vi...@gmail.com wrote: He's doing an extra effort... just compile it as you would normally and you will have the debug symbols. On Tue, Sep 22, 2009 at 8:11 PM, Diego Viola diego.vi...@gmail.comwrote: Then why is Tihomir trying to compile with debug symbols? On Tue, Sep 22, 2009 at 8:00 PM, Brian West br...@freeswitch.org wrote: yes On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: Doesn't FS already compiles with debug symbols by default? ___ 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] SUBSCRIBE and NOTIFY
On Tue, Sep 22, 2009 at 12:20 PM, Rudá Cunha r...@ruda.com.br wrote: I'm having to configure FreeSWITCH. Download version 1.0.4 and I am accessing with the users 1000 and 1001. I register, make the connection. But I'm trying to see to see who is connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I connect the other User does not receive the information that I connected (I sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected to another user I'm not, sometimes you work. What are you using to see who is connected? -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] Can this be done in FreeSWITCH?
Hi Michael, The DID is transported via SIP to the router. From the router to the POTS provider, it's PRI. On Wed, Sep 23, 2009 at 12:34 AM, Michael Collins m...@freeswitch.orgwrote: How is the DID transported? SIP, PRI, analog DID trunks? -MC On Mon, Sep 21, 2009 at 9:17 PM, Francis Vidal francisv.l...@gmail.comwrote: Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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 this be done in FreeSWITCH?
Yes, this is the desired outcome. I was planning of using FreeSWITCH + MySQL to do this. How do I do this inline? On Wed, Sep 23, 2009 at 12:49 AM, Tihomir Culjaga tculj...@gmail.comwrote: so, you say ... CallingParty = AS5300 A: aNum B: didNum AS5300 = PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. ___ 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] Multitenancy
Hello all, How do I configure multi tenant in FS? For example, I want some users to be able to register only with their own domain. Ie: Users: 1000-1010 Domain: foo.org Users: 2000-2010 Domain: bar.org But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work with foo.org. Any ideas how to do 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
Re: [Freeswitch-users] Multitenancy
I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to bar.org, and I want both of those domains to have their own dialplan/context. On Wed, Sep 23, 2009 at 4:57 AM, Diego Viola diego.vi...@gmail.com wrote: Hello all, How do I configure multi tenant in FS? For example, I want some users to be able to register only with their own domain. Ie: Users: 1000-1010 Domain: foo.org Users: 2000-2010 Domain: bar.org But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work with foo.org. Any ideas how to do 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
Re: [Freeswitch-users] Multitenancy
I know I could create different domains on the directory but how do I tell a user to belong to a specific domain? On Wed, Sep 23, 2009 at 5:11 AM, Diego Viola diego.vi...@gmail.com wrote: I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to bar.org, and I want both of those domains to have their own dialplan/context. On Wed, Sep 23, 2009 at 4:57 AM, Diego Viola diego.vi...@gmail.comwrote: Hello all, How do I configure multi tenant in FS? For example, I want some users to be able to register only with their own domain. Ie: Users: 1000-1010 Domain: foo.org Users: 2000-2010 Domain: bar.org But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work with foo.org. Any ideas how to do 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
Re: [Freeswitch-users] Multitenancy
Then setup two domains in your directory and setup proper DNS its really just that simple. /b On Sep 23, 2009, at 12:11 AM, Diego Viola wrote: I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to bar.org, and I want both of those domains to have their own dialplan/context. ___ 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] Multitenancy
Oh nvm I think I got it =D On Wed, Sep 23, 2009 at 5:22 AM, Diego Viola diego.vi...@gmail.com wrote: I know I could create different domains on the directory but how do I tell a user to belong to a specific domain? On Wed, Sep 23, 2009 at 5:11 AM, Diego Viola diego.vi...@gmail.comwrote: I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to bar.org, and I want both of those domains to have their own dialplan/context. On Wed, Sep 23, 2009 at 4:57 AM, Diego Viola diego.vi...@gmail.comwrote: Hello all, How do I configure multi tenant in FS? For example, I want some users to be able to register only with their own domain. Ie: Users: 1000-1010 Domain: foo.org Users: 2000-2010 Domain: bar.org But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work with foo.org. Any ideas how to do 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
Re: [Freeswitch-users] mod_conference performance (Brian West)
It was one big conference. Robert ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mod_perl $session in not hangup
Thanks for your replay. I don't know what is latest trunk. Is it latest
version? I'm using freeswitch 1.0.4.
On Tue, Sep 22, 2009 at 8:09 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
The reason is you cannot complete the hangup until the script exits.
On the bright side, if you update to latest trunk it will probably work
more how you want it to
because a recent change will make this possible.
On Tue, Sep 22, 2009 at 4:30 AM, lakshmanan lakindi...@gmail.com wrote:
Hi all, I've the following mod_perl program to execute when I call to an
extension (say 777).
I use twinkle as a soft phone, to make calls.
#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
$session-answer();
if($session-ready())
{
my $uuid=$session-getVariable(uuid);
freeswitch::consoleLog(INFO,UUID is $uuid\n);
freeswitch::consoleLog(INFO,Session is answered\n);
$session-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav);
my $dtmf = $session-getDigits(4,, 5000);
freeswitch::consoleLog(INFO,I received $dtmf\n);
$session-hangup(NORMAL_CLEARING);
sleep(5);
# Some other statements.
}
return 1;
Everything is fine.
After executing $session-hangup, I got NORMAL_CLEARING in my freeswitch
console. But in my soft phone, still the channel is active for 5 seconds.
The call got ended only after the 5 seconds sleep.
But if I create my own session like
my $session=new
freeswitch::Session(user/1000);
and I say $session-hangup(), it got terminated.
I wanted to know why there is such difference?? or am I wrong??
Please clarify me.
--
View this message in context:
http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Anthony Minessale II
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