Re: [Freeswitch-users] Hi Guys
Question: Will the call flow look like this (above was not very clear): web -- FS -- Cantata -- PSTN (via TDM circuits) Or are you trying to replace the Cantata? SDR ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] UUID of the newly originated call?
I've tried with origination_uuid. First, I tried with SIP and my program executes successfully as what I expected. This program initiates a new call when a call comes and let the new call to eavesdrop the landed call. When, I experimented with PRI(openzap), I'm facing the following error. And, I am unable to make any calls(even from CLI). It is reporting the same error for the subsequent calls. Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9] Full log is in http://pastebin.freeswitch.org/10780 My script is here, http://pastebin.freeswitch.org/10781 What is this error for and how to avoid this? Is there any other way to get the uuid of the originated call except explicitly defining(origination_uuid).?! -- Regards, Nagalenoj H. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
Sort of. I have different error now: 2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal-ipv6/1...@franta.openstage.net! 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup sofia/internal-ipv6/sip:1...@[2000:2::1002]:5062 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Here is the complete log: http://pastebin.freeswitch.org/10782 Yes. Are you suggesting it didn't 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] Hi Guys
Well ... the question is pretty generic. But based on these assumptions: -- no media (bypass media) -- routing done via XML dialplan Something along the lines of a quad core machine with 4 gigs of ram would be overkill for 692 calls. Things to remember: -- the more cores the better (FS is heavily threaded) -- the more memory the better -- 64 bit is way better than 32 bit SDR ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
ineya ineya ine...@gmail.com wrote: Sort of. I have different error now: 2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal-ipv6/1...@franta.openstage.net! 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup sofia/internal-ipv6/sip:1...@[2000:2::1002]:5062 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] I suspect the codec negotiation. Make sure that both ends are offering a common codec. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
simple:
action application=bridge data=h323/${number}/
if fs not registered on gk then data=h323/${numb...@xxx.xxx.xxx.xxx.
TC
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/PointDevice 14
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
thread
TC0xb6eb60a0 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external
thread
TC0xb6ebafa8 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/Modal 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
thread
TC0xb6eba910 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TCSegmentation fault (core dumped)
TCtculj...@subzero:~/freeswitch-trunk$
TC
TCpls check: http://pastebin.freeswitch.org/10769
look strange, what version of libpt/h323plus you use and freeswitch itself
?
TC
TC
I was using latest libpt.so.2.7-beta1.
Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...)
and FS is crashing on every call :P .. regardless if it is inbound or
outbound...
FreeSWITCH Version 1.0.trunk (15079M)
H323Plus is from cvs
so, what i did is:
create a directory e.g. h323
mkdir -p ~/h323
cd ~/h323
svn co
http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6
export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
export LD_LIBRARY_PATH=/usr/local/lib
cd ptlib-2.6
./configure
make
sudo make install
cd ~/h323
cvs -d:pserver:anonym...@h323plus.cvs.sourceforge.net:/cvsroot/h323plus
checkout h323plus
export PTLIBDIR=~/h323plus/ptlib
cd h323plus
./configure
make
sudo make install
assuming you have FS src in your home
cd ~/freeswitch-trunk
make mod_h323-clean
make mod_h323
sudo make mod_h323-install
cd /usr/local/freeswitch/lib/
sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6
start FS and load mod_h323
Please, can you advice what exact revisions of ptlib you are using so i can
do svn so -r xxx, also what exact revision of freeswitch and H323Plus you
are using ?
Now with ptlib-2.6-beta6 can't even.
T.
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Re: [Freeswitch-users] Qustion about INFO messages after Connect/Answer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Brian, hi Mike, On 20.10.2009 18:41, Brian West wrote: Or just set the var to what you want it to say? Yes, understood and it works so far. This means that I must enhance my dialplan to set this new variable to preserve old behaviour. No big deal, but at least I have to know it. /b On Oct 20, 2009, at 11:19 AM, Michael Collins wrote: Under what conditions did you see unknown? I'm wondering if the user can just pick a default other than unknown if he wants something else to be displayed. I get it for internal calls from Snom to Snom. It seems to be the default configuration. The sip flow shows two INFO messages sent from FS to caller after callee picked up. The first INFO messages set the callee's name to unknown on caller's side. The second changed it back to callee's number. Maybe there is a plan behind it ... by now it is simply increasing the sip signalling load. Any ideas for what the first INFO message is? regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK3sXF4tZeNddg3dwRAshzAJ99Jsp/RNtndeulae80pvHPqC9YHACghFxT y0JZzsSKrGyPXTnPypy+qqQ= =jNtK -END PGP SIGNATURE- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sched_api doesn't get launched
I can't seem to find the right thing to use in mod_java to execute api commands, only api_after_bridge 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1688...@192.168.1.66! # # A fatal error has been detected by the Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624 # # JRE version: 6.0_16-b01 # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 ) # Problematic frame: # C [libc.so.6+0x6f480] strcpy+0x10 # # An error report file with more information is saved as: # /usr/local/freeswitch/bin/hs_err_pid1927.log 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid Application sched_api 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup sofia/internal/1688...@192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # The crash happened outside the Java Virtual Machine in native code. # See problematic frame for where to report the bug. On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.org wrote: On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang red.rain.se...@gmail.comwrote: So how would you trigger it from a script dialplan? The only time it seemed to work is when I did setVariable(api_after_bridge, sched_api blah blah blah); but then it gets executed after the channel's been teared down. I thought api_after_bridge means right after the call gets connected. I need something to execute an api command right before or right after the call gets bridged. api_after_bridge is a channel variable, so using setVariable works just fine. If you need to sched_api is an API only. Check these out: http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) Remember, if the method you are using isn't found in the dial plan tools then it isn't a dial plan application. Make sure it's on the list: http://wiki.freeswitch.org/wiki/Mod_dptools On the other hand, API commands are listed here: http://wiki.freeswitch.org/wiki/Mod_commands dptools require a session object, api commands require an api object... -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC TC TCI was using latest libpt.so.2.7-beta1. TC TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...) TCand FS is crashing on every call :P .. regardless if it is inbound or TCoutbound... http://www.opalvoip.org/ first link into Lalande Stable 5 Released. On some version of cvs ptlib i get crash on module loading:) TC TCFreeSWITCH Version 1.0.trunk (15079M) hm, i don't test it on trunk, may be there some isues, try get stack backtrace from core file to see where it crash. I use 1.0.4 TCH323Plus is from cvs it's ok. TC TCso, what i did is: TC TC TCcreate a directory e.g. h323 TC TCmkdir -p ~/h323 TCcd ~/h323 TC TC svn co TChttp://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6 TC TC export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig TC export LD_LIBRARY_PATH=/usr/local/lib TC TCcd ptlib-2.6 TC TC ./configure TC make TC sudo make install TC TC TC TCcd ~/h323 TC TC cvs -d:pserver:anonym...@h323plus.cvs.sourceforge.net:/cvsroot/h323plus TCcheckout h323plus TC TC export PTLIBDIR=~/h323plus/ptlib TC TC TCcd h323plus TC TC ./configure TC make TC sudo make install TC TC TC TCassuming you have FS src in your home TC TCcd ~/freeswitch-trunk TC TC make mod_h323-clean TC make mod_h323 TC sudo make mod_h323-install TC TC TC TCcd /usr/local/freeswitch/lib/ TC TC sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6 TC TC TC TC TCstart FS and load mod_h323 TC TC TC TC TC TCPlease, can you advice what exact revisions of ptlib you are using so i can TCdo svn so -r xxx, also what exact revision of freeswitch and H323Plus you TCare using ? TC TC TC TC TC TCNow with ptlib-2.6-beta6 can't even. TC TC TCT. TC C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call custom variable in condition
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I want
to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS} XML
Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
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Re: [Freeswitch-users] Call custom variable in condition
Ahmed,
you can't use variables set by set application within a condition,
though it doesn't make sense. wondering if there is any logic behind
this or it's just a simple missing feature. anyone?
-m
Ahmed Munir wrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and
I want to use it in condition, as I'm listing down the configuration
below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer
data=${AUTHENTICATION_STATUS} XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do
implement it(Note: I don't want to call script). And one more thing to
ask how can I transfer the values within the same context?
--
Regards,
Ahmed Munir
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Re: [Freeswitch-users] Call custom variable in condition
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS} XML
Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=${AUTHENTICATION_STATUS} expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
here is one of my dialplan. I'm using execute_extension but it is quite the
same...
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
/condition
/extension
extension name=doRedirect
condition field=destination_number expression=^doRedirect$/
condition field=${authResult} expression=^0$|^60$
action application=log data=INFO
RADIUS auth OK!!!' ##\n/
action application=redirect data=${red_contact}/
anti-action application=log data=INFO
RADIUS auth NOK!! ##\n/
anti-action application=respond data=403 Forbidden/
/condition
/extension
On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
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[Freeswitch-users] NOT in dialplan expression
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to work in FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/21 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC TC TCI was using latest libpt.so.2.7-beta1. TC TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...) TCand FS is crashing on every call :P .. regardless if it is inbound or TCoutbound... http://www.opalvoip.org/ first link into Lalande Stable 5 Released. On some version of cvs ptlib i get crash on module loading:) TC TCFreeSWITCH Version 1.0.trunk (15079M) hm, i don't test it on trunk, may be there some isues, try get stack backtrace from core file to see where it crash. I use 1.0.4 module load crash: http://pastebin.freeswitch.org/10783 FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784 now, the only different thing is FS trunk ... :P ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] NOT in dialplan expression
it depends of what you are trying to acheave one approach is with regex
check this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex
you can set a different variable and have it true or false ... than you can
compare for false state...
well .. it is up to you ...
T.
On Wed, Oct 21, 2009 at 1:34 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to work in
FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
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Re: [Freeswitch-users] NOT in dialplan expression
Hi,
Negating is done with [^...] in a regex, so 'not 1' is matched with:
/^[^1]$/
If you want to match on a longer sequence, you can do that with
negative lookahead, for example 'not 123' can be matched like this:
/^(?!123$)\d{3}$/
regards,
Leon
On Oct 21, 2009, at 1:34 PM, Mark Campbell-Smith wrote:
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to
work in FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
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Re: [Freeswitch-users] can't dial from IPv6 to IPv6
Codecs are fine. I spent much time experimenting with codecs and completely missed, that freeswitch is modifiyng the SDP record. When phone A is making a call the SDP contains candidate media attributes: a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host But when freeswitch makes the INVITE on phone B, these 2 are missing and phone is looking for it, so the INVITE gets rejected by phone with 448 Not acceptable here So the question is, how can I make the freeswitch to pass these candidate media attributes? On Wed, Oct 21, 2009 at 8:58 AM, Jason White ja...@jasonjgw.net wrote: I suspect the codec negotiation. Make sure that both ends are offering a common codec. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC2009/10/21 Georgiewskiy Yuriy bottle...@icf.org.ru TC TC On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote TC freeswitch-us...@lists.fre...: TC TC TC TC TC TC TC TC TC TCI was using latest libpt.so.2.7-beta1. TC TC TC TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you TC mentioned...) TC TCand FS is crashing on every call :P .. regardless if it is inbound or TC TCoutbound... TC TC http://www.opalvoip.org/ first link into Lalande Stable 5 Released. TC On some version of cvs ptlib i get crash on module loading:) TC TC TC TC TCFreeSWITCH Version 1.0.trunk (15079M) TC TC hm, i don't test it on trunk, may be there some isues, try get stack TC backtrace from core file to TC see where it crash. I use 1.0.4 TC TC TC TC TCmodule load crash: http://pastebin.freeswitch.org/10783 TCFreeSWITCH backtrace: http://pastebin.freeswitch.org/10784 TC TC TCnow, the only different thing is FS trunk ... i have no trunk at this time and cannot test it, have you enabled crash-protection in fs? C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ FreeSWITCH-users mailing list FreeSWITCH-users@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] check sip client availability
Hi, Is there any API to tell freeswitch to send a SIP OPTION message to check the availability of a SIP client? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
What are you using to make this call? /b On Oct 21, 2009, at 6:58 AM, ineya ineya wrote: Codecs are fine. I spent much time experimenting with codecs and completely missed, that freeswitch is modifiyng the SDP record. When phone A is making a call the SDP contains candidate media attributes: a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host But when freeswitch makes the INVITE on phone B, these 2 are missing and phone is looking for it, so the INVITE gets rejected by phone with 448 Not acceptable here So the question is, how can I make the freeswitch to pass these candidate media attributes? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
Your SVN Account will be done soon and the directory in endpoints is already created for you to start importing your work. Thanks, /b On Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote: On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre ...: TC2009/10/21 Georgiewskiy Yuriy bottle...@icf.org.ru TC TC On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote TC freeswitch-us...@lists.fre...: TC TC TC TC TC TC TC TC TC TCI was using latest libpt.so.2.7-beta1. TC TC TC TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you TC mentioned...) TC TCand FS is crashing on every call :P .. regardless if it is inbound or TC TCoutbound... TC TC http://www.opalvoip.org/ first link into Lalande Stable 5 Released. TC On some version of cvs ptlib i get crash on module loading:) TC TC TC TC TCFreeSWITCH Version 1.0.trunk (15079M) TC TC hm, i don't test it on trunk, may be there some isues, try get stack TC backtrace from core file to TC see where it crash. I use 1.0.4 TC TC TC TC TCmodule load crash: http://pastebin.freeswitch.org/10783 TCFreeSWITCH backtrace: http://pastebin.freeswitch.org/10784 TC TC TCnow, the only different thing is FS trunk ... i have no trunk at this time and cannot test it, have you enabled crash-protection in fs? C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129- RIPE___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-21 09:26 -0500, Brian West wrote freeswitch-us...@lists.freeswit...: ок. BWYour SVN Account will be done soon and the directory in endpoints is BWalready created for you to start importing your work. BW BWThanks, BW BW/b BW BWOn Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote: BW BW On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre BW ...: BW BW TC2009/10/21 Georgiewskiy Yuriy bottle...@icf.org.ru BW TC BW TC On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote BW TC freeswitch-us...@lists.fre...: BW TC BW TC BW TC TC BW TC TC BW TC TC BW TC TCI was using latest libpt.so.2.7-beta1. BW TC TC BW TC TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you BW TC mentioned...) BW TC TCand FS is crashing on every call :P .. regardless if it is BW inbound or BW TC TCoutbound... BW TC BW TC http://www.opalvoip.org/ first link into Lalande Stable 5 BW Released. BW TC On some version of cvs ptlib i get crash on module loading:) BW TC BW TC TC BW TC TCFreeSWITCH Version 1.0.trunk (15079M) BW TC BW TC hm, i don't test it on trunk, may be there some isues, try get BW stack BW TC backtrace from core file to BW TC see where it crash. I use 1.0.4 BW TC BW TC BW TC BW TC BW TCmodule load crash: http://pastebin.freeswitch.org/10783 BW TCFreeSWITCH backtrace: http://pastebin.freeswitch.org/10784 BW TC BW TC BW TCnow, the only different thing is FS trunk ... BW BW i have no trunk at this time and cannot test it, have you enabled BW crash-protection in fs? BW BW C уважением With Best Regards BW Георгиевский Юрий.Georgiewskiy Yuriy BW +7 4872 711666+7 4872 711666 BW факс +7 4872 711143 fax +7 4872 711143 BW Компания ООО Ай Ти Сервис IT Service Ltd BW http://nkoort.ru http://nkoort.ru BW JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru BW YG129-RIPEYG129- BW RIPE___ BW FreeSWITCH-users mailing list BW FreeSWITCH-users@lists.freeswitch.org BW http://lists.freeswitch.org/mailman/listinfo/freeswitch-users BW UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- BW users BW http://www.freeswitch.org BW BW BW___ BWFreeSWITCH-users mailing list BWFreeSWITCH-users@lists.freeswitch.org BWhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users BWUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users BWhttp://www.freeswitch.org BW C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ FreeSWITCH-users mailing list FreeSWITCH-users@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] NOT in dialplan expression
You can also look at using anti-action rather than action after the condition.
condition == if
action == then
anti-action == else
On Wed, Oct 21, 2009 at 6:34 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to work in
FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
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Re: [Freeswitch-users] check sip client availability
This applies only to clients that are detected as nat, but maybe: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping This is applied to the profile, not an app or a var setting. On Wed, Oct 21, 2009 at 8:26 AM, Woody Dickson woodydick...@gmail.com wrote: Hi, Is there any API to tell freeswitch to send a SIP OPTION message to check the availability of a SIP client? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-21 16:43 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC TC TC TCmodule load crash: http://pastebin.freeswitch.org/10783 TC TCFreeSWITCH backtrace: http://pastebin.freeswitch.org/10784 TC TC TC TC TC TCnow, the only different thing is FS trunk ... TC TC i have no trunk at this time and cannot test it, have you enabled TC crash-protection in fs? TC TC TCOk i moved back FS to 1.0.4 and i'm back on the first issues, TC TC1. no audio after i answer the call (call flow is H323 = FS = SIP).. i TChear the ringback on the H323 side bit when i answer the call, nothing! TCFS = 10.4.62.7 TCSIP phone = 10.4.62.89 TCH323 endpoint = 10.1.14.153 TC 53.696259 10.1.14.153 - 10.4.62.7H.225.0 CS: setup TC 53.70541510.4.62.7 - 10.1.14.153 H.225.0 CS: callProceeding TC 53.71636810.4.62.7 - 10.4.62.89 SIP/SDP Request: INVITE TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89;transport=udp, with session TCdescription TC 53.71769710.4.62.7 - 10.1.14.153 H.225.0 CS: alerting TC 53.72013210.4.62.7 - 10.1.14.153 H.225.0 CS: facility TC 53.72545310.4.62.7 - 10.1.14.153 H.245 terminalCapabilitySet TC 53.72595710.4.62.7 - 10.1.14.153 H.245 masterSlaveDetermination TC 53.728129 10.4.62.89 - 10.4.62.7SIP Status: 100 Trying TC 53.741112 10.4.62.89 - 10.4.62.7SIP Status: 180 Ringing TC 53.743199 10.1.14.153 - 10.4.62.7H.245 terminalCapabilitySet TC 53.744221 10.1.14.153 - 10.4.62.7H.245 masterSlaveDetermination TC 53.74944310.4.62.7 - 10.1.14.153 H.245 terminalCapabilitySetAck TC 53.75162410.4.62.7 - 10.1.14.153 H.245 masterSlaveDeterminationAck TC 53.758710 10.1.14.153 - 10.4.62.7H.245 terminalCapabilitySetAck TC 53.761241 10.1.14.153 - 10.4.62.7H.245 masterSlaveDeterminationAck TC 53.76391910.4.62.7 - 10.1.14.153 H.245 openLogicalChannel (g711A) TC#3712: OLC found 10.1.14.153/10.4.62.7/101 TC 53.777464 10.1.14.153 - 10.4.62.7H.245 openLogicalChannelAck TC 53.79986410.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1305, Time=240, Mark TC 53.82894010.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1306, Time=480 TC 53.85918010.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1307, Time=720 TC 53.88937910.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1308, Time=960 TC 53.91961110.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1309, Time=1200 TC 53.94983310.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1310, Time=1440 TC 53.97896410.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1311, Time=1680 TC 54.00921810.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1312, Time=1920 TC 54.03942410.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1313, Time=2160 TC 54.06964010.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1314, Time=2400 TC 54.09979510.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1315, Time=2640 TC 54.12906410.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1316, Time=2880 TC TC-- snip this is a ringback sent from FS TC= H323endpoint TC TC 60.18907410.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1518, Time=51360 TC 60.21919710.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1519, Time=51600 TC 60.24945110.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1520, Time=51840 TC 60.27968410.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1521, Time=52080 TC 60.30968910.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1522, Time=52320 TC 60.33881210.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1523, Time=52560 TC 60.36899710.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1524, Time=52800 TC 60.39923110.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1525, Time=53040 TC 60.42944510.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1526, Time=53280 TC 60.45967910.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1527, Time=53520 TC 60.48891710.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1528, Time=53760 TC 60.494975 10.4.62.89 - 10.4.62.7SIP/SDP Status: 200 OK, with session TCdescription TC 60.49547810.4.62.7 - 10.4.62.89 SIP Request: ACK TCsip:1...@10.4.62.89sip%3a1...@10.4.62.89 TC;transport=udp TC 60.49611410.4.62.7 - 10.4.62.7RTP Unknown RTP version 1 TC 60.51946710.4.62.7 - 10.1.14.153 RTP PT=ITU-T G.711 PCMA, TCSSRC=0xBE241F, Seq=1529, Time=54000 TC 60.52057410.4.62.7 - 10.1.14.153 H.225.0 CS: connect TC 60.531284 10.1.14.153 - 10.4.62.7H.245 openLogicalChannel (g711A) TC#4045: OLC found
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
the phone is called OpenStage On Wed, Oct 21, 2009 at 4:25 PM, Brian West br...@freeswitch.org wrote: What are you using to make this call? /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sched_api doesn't get launched
Judging by this error I would assume that you're still calling sched_api as a Dialplan application and not as an FS API command. You need to figure out how to create an API obj in java and call sched_api from that object. -MC Sent from my iPhone On Oct 21, 2009, at 2:44 AM, Henry Huang red.rain.se...@gmail.com wrote: I can't seem to find the right thing to use in mod_java to execute api commands, only api_after_bridge 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1688...@192.168.1.66! # # A fatal error has been detected by the Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624 # # JRE version: 6.0_16-b01 # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 ) # Problematic frame: # C [libc.so.6+0x6f480] strcpy+0x10 # # An error report file with more information is saved as: # /usr/local/freeswitch/bin/hs_err_pid1927.log 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid Application sched_api 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup sofia/internal/1688...@192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # The crash happened outside the Java Virtual Machine in native code. # See problematic frame for where to report the bug. On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.org wrote: On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang red.rain.se...@gmail.com wrote: So how would you trigger it from a script dialplan? The only time it seemed to work is when I did setVariable(api_after_bridge, sched_api blah blah blah); but then it gets executed after the channel's been teared down. I thought api_after_bridge means right after the call gets connected. I need something to execute an api command right before or right after the call gets bridged. api_after_bridge is a channel variable, so using setVariable works just fine. If you need to sched_api is an API only. Check these out: http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) Remember, if the method you are using isn't found in the dial plan tools then it isn't a dial plan application. Make sure it's on the list: http://wiki.freeswitch.org/wiki/Mod_dptools On the other hand, API commands are listed here: http://wiki.freeswitch.org/wiki/Mod_commands dptools require a session object, api commands require an api object... -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153
TC2. hangup from sip side doesn't release the h323 leg (now the difference
is
TCthat FS is not complaining about thread mismatch ant it looks clean but
FS
TCdoesn't send any releasecomplete message... strange)
TC3. coredumps when i place outgoing calls
btw,
TC 70.552157 10.1.14.153 - 10.4.62.7H.245 endSessionCommand
TC 70.552401 10.1.14.153 - 10.4.62.7H.225.0 CS: releaseComplete
TC 70.55398110.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
TC 70.55448810.4.62.7 - 10.1.14.153 H.245 endSessionCommand
TC 70.55505610.4.62.7 - 10.1.14.153 H.225.0 CS: releaseComplete
it send, now i have no way to test h323-sip transit, i will have it
tomorow.
sip-h323 for me work fine now, give backtrace from code dump of 1.0.4
where it die?
this endSession is when i hangup from H232 side as well :P ... if i don't
hangup on H323 side the H323 leg is not released. Pls chec the time the
packets were sent ...
Here i hangup on the SIP Phone:
68.374916 10.4.62.89 - 10.4.62.7SIP Request: BYE
sip:mod_so...@10.4.62.7:5060
68.37534210.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
68.37562010.4.62.7 - 10.4.62.89 SIP Status: 200 OK
2 sec delay
Here i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
first here )
70.552157 10.1.14.153 - 10.4.62.7H.245 endSessionCommand
70.552401 10.1.14.153 - 10.4.62.7H.225.0 CS: releaseComplete
FS just acknowlages it here:
70.55448810.4.62.7 - 10.1.14.153 H.245 endSessionCommand
70.55505610.4.62.7 - 10.1.14.153 H.225.0 CS: releaseComplete
I have enabled crash-protection and when i do SIP = H323 call it doesn't
generate coredumps... it is just this thread that is crashing ... pls check
the log downbelow:
Dialplan: sofia/internal/1...@singtel Regex (FAIL) [EMERGENCY]
destination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
Dialplan: sofia/internal/1...@singtel parsing [default-SPECIAL_SERVICES]
continue=false
Dialplan: sofia/internal/1...@singtel Regex (FAIL) [SPECIAL_SERVICES]
destination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
Dialplan: sofia/internal/1...@singtel parsing [default-ENYTHING_ELSE]
continue=false
Dialplan: sofia/internal/1...@singtel Regex (PASS) [ENYTHING_ELSE]
destination_number(05492122) =~
/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
Dialplan: sofia/internal/1...@singtel Action
set(effective_caller_id_number=1001282122)
Dialplan: sofia/internal/1...@singtel Action set(NCX_IP=10.4.4.254)
Dialplan: sofia/internal/1...@singtel Action set(call_timeout=30)
Dialplan: sofia/internal/1...@singtel Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1...@singtel Action set(bypass_media=false)
Dialplan: sofia/internal/1...@singtel Action set(proxy_media=true)
Dialplan: sofia/internal/1...@singtel Action bridge(h323/05492...@${ncx_ip})
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1...@singtel) State Change CS_ROUTING - CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1...@singtel [BREAK]
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1...@singtel) State ROUTING going to sleep
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1...@singtel) Running State Change CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1...@singtel) State EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
sofia/internal/1...@singtel SOFIA EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1...@singtel Standard EXECUTE
EXECUTE sofia/internal/1...@singtel set(open=true)
2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [open]=[true]
EXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
EXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE sofia/internal/1...@singtelset(effective_caller_id_number=1001282122)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [effective_caller_id_number]=[1001282122]
EXECUTE sofia/internal/1...@singtel set(NCX_IP=10.4.4.254)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [NCX_IP]=[10.4.4.254]
EXECUTE sofia/internal/1...@singtel set(call_timeout=30)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [call_timeout]=[30]
EXECUTE sofia/internal/1...@singtel set(hangup_after_bridge=true)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1...@singtel set(bypass_media=false)
2009-10-21
[Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
Sometimes if I stop (using ... command) and then restart freeswitch (using ./freeswitch command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Stop/Restart of Freeswitch Causes Crash
You should debug FreeSWITCH, check this out: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch And then open a Jira. Diego On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards jerry.richa...@teotech.com wrote: Sometimes if I stop (using ... command) and then restart freeswitch (using ./freeswitch command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Stop/Restart of Freeswitch Causes Crash
That problem is fixed if you update to SVN trunk or latest snapshot. On Wed, Oct 21, 2009 at 11:11 AM, Jerry Richards jerry.richa...@teotech.com wrote: Sometimes if I stop (using ... command) and then restart freeswitch (using ./freeswitch command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call custom variable in condition
It not only makes sense it's well documented on the wiki page.
The set line is not happening right when it's encountered, the set line is
copied into the channel and executed later after the whole dialplan is
parsed. The dialplan is a pre-processor not a runtime engine.
Here is a new feature in pre-1.0.5 (svn trunk)
Some applications like set can now be executed within the dialplan but you
should use it sparingly.
action application=set data=testing=true inline=true/
The inline=true makes it execute inside the dialplan and it's never copied
into your resulting extension because it's executed immediately.
On Wed, Oct 21, 2009 at 6:13 AM, Tihomir Culjaga tculj...@gmail.com wrote:
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=${AUTHENTICATION_STATUS} expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
here is one of my dialplan. I'm using execute_extension but it is quite the
same...
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
/condition
/extension
extension name=doRedirect
condition field=destination_number expression=^doRedirect$/
condition field=${authResult} expression=^0$|^60$
action application=log data=INFO
RADIUS auth OK!!!' ##\n/
action application=redirect data=${red_contact}/
anti-action application=log data=INFO
RADIUS auth NOK!! ##\n/
anti-action application=respond data=403 Forbidden/
/condition
/extension
On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
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--
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] Hi Guys
Thank you guysEd On Wed, Oct 21, 2009 at 2:49 AM, Shelby Ramsey sicfsl...@gmail.com wrote: Well ... the question is pretty generic. But based on these assumptions: -- no media (bypass media) -- routing done via XML dialplan Something along the lines of a quad core machine with 4 gigs of ram would be overkill for 692 calls. Things to remember: -- the more cores the better (FS is heavily threaded) -- the more memory the better -- 64 bit is way better than 32 bit SDR ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Stop/Restart of Freeswitch Causes Crash
no you shouldnt we don't take jiras about release revisions only trunk. On Wed, Oct 21, 2009 at 11:31 AM, Diego Viola diego.vi...@gmail.com wrote: You should debug FreeSWITCH, check this out: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch And then open a Jira. Diego On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards jerry.richa...@teotech.com wrote: Sometimes if I stop (using ... command) and then restart freeswitch (using ./freeswitch command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Is background sound|music during call possible?
Hello How can I add background music that will play during call? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Calls dropping on SIP timer expiry due to UPDATE's being ignored.
Hi, Hope someone knows how I am able to get around this one. Here goes... Did an upgrade to trunk (from a July vintage build) last week and noticed calls out to a provider were now failing after about 30 seconds or so - post answer. Tried latest (15183) - same thing. Analysing, I see that I have multiple UPDATE messages now being sent to the provider, but no response being sent back to FS. So FS times out and eventually kills the call. Interestingly, it only drops the A-leg; the B-leg remains up till the B party hangs up. I cant recall seeing these UPDATE messages before... The intent of the UPDATE seems to be to send the callee name number to the B-leg. If its the provider's sip stack that's broken w.r.t. handling UPDATE - is there any way to get around it by doing something in my config to ensure these UPDATE's are not 'triggered' ? Some traces below. Any suggestions welcomed... Best Regards Keith Pretoria, South Africa. -- send 1048 bytes to udp/[196.10.11.12]:5060 at 13:24:04.249269: INVITE sip:27835551...@196.10.11.12 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m Max-Forwards: 67 From: Keith PhoneADSL sip:878050...@10.17.10.10;tag=Upa3NvXpBB1eF To: sip:27835551...@196.10.11.12 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: sip:gw+vp...@10.17.10.10:5060;transport=udp;gw=vprov User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-Actually-Support: UPDATE Remote-Party-ID: Keith PhoneADSL sip:878050...@10.17.10.10;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2009-10-21 15:24:04.248448 [DEBUG] sofia.c:3493 Channel sofia/vvrf/2783555 entering state [calling][0] recv 601 bytes from udp/[196.10.11.12]:5060 at 13:24:04.307690: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: Keith PhoneADSL sip:878050...@10.17.10.10;tag=Upa3NvXpBB1eF To: sip:27835551...@196.10.11.12;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: vprov C5CM sip:196.10.11.12:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: Keith PhoneADSL sip:878050...@10.17.10.10;party=calling;screen=yes;privacy=off Content-Length: 0 recv 879 bytes from udp/[196.10.11.12]:5060 at 13:24:08.508162: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: Keith PhoneADSL sip:878050...@10.17.10.10;tag=Upa3NvXpBB1eF To: sip:27835551...@196.10.11.12;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: vprov C5CM sip:196.10.11.12:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: Keith PhoneADSL sip:878050...@10.17.10.10;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 233 v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3493 Channel sofia/vvrf/2783555 entering state [proceeding][183] 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3500 Remote SDP: v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-10-21 15:24:08.507506 [DEBUG] sofia_glue.c:3144 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2009-10-21 15:24:08.508561
Re: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
Oh ok, I didn't knew that Anthony, sorry. Do as Anthony said, update to trunk :). On Wed, Oct 21, 2009 at 4:54 PM, Anthony Minessale anthony.miness...@gmail.com wrote: no you shouldnt we don't take jiras about release revisions only trunk. On Wed, Oct 21, 2009 at 11:31 AM, Diego Viola diego.vi...@gmail.com wrote: You should debug FreeSWITCH, check this out: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch And then open a Jira. Diego On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards jerry.richa...@teotech.com wrote: Sometimes if I stop (using ... command) and then restart freeswitch (using ./freeswitch command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call custom variable in condition
On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale anthony.miness...@gmail.com wrote: It not only makes sense it's well documented on the wiki page. The set line is not happening right when it's encountered, the set line is copied into the channel and executed later after the whole dialplan is parsed. The dialplan is a pre-processor not a runtime engine. Here is a new feature in pre-1.0.5 (svn trunk) Some applications like set can now be executed within the dialplan but you should use it sparingly. action application=set data=testing=true inline=true/ I'm getting ready to document this feature. For the sake of edification, why is it best to use this sparingly, other than wide-spread use making dialplans all cluttered? -MC The inline=true makes it execute inside the dialplan and it's never copied into your resulting extension because it's executed immediately. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call custom variable in condition
you should not abuse it is all i mean, we have measures to limit what apps you can use in this manner but usually requiring a more complicated dialplan is a hint you are doing something wrong ;) On Wed, Oct 21, 2009 at 12:02 PM, Michael Collins m...@freeswitch.orgwrote: On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale anthony.miness...@gmail.com wrote: It not only makes sense it's well documented on the wiki page. The set line is not happening right when it's encountered, the set line is copied into the channel and executed later after the whole dialplan is parsed. The dialplan is a pre-processor not a runtime engine. Here is a new feature in pre-1.0.5 (svn trunk) Some applications like set can now be executed within the dialplan but you should use it sparingly. action application=set data=testing=true inline=true/ I'm getting ready to document this feature. For the sake of edification, why is it best to use this sparingly, other than wide-spread use making dialplans all cluttered? -MC The inline=true makes it execute inside the dialplan and it's never copied into your resulting extension because it's executed immediately. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Is background sound|music during call possible?
uuid_displace with mux option. /b On Oct 21, 2009, at 6:11 AM, Joey Carter wrote: Hello How can I add background music that will play during call? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls dropping on SIP timer expiry due to UPDATE's being ignored.
This will be fixed soon. Watch SVN. /b On Oct 21, 2009, at 11:45 AM, Keith Laaks wrote: Hi, ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
This appears to be some sort of ice implementation? We don't support sip ice at this time. Mike On Oct 21, 2009, at 7:58 AM, ineya ineya wrote: Codecs are fine. I spent much time experimenting with codecs and completely missed, that freeswitch is modifiyng the SDP record. When phone A is making a call the SDP contains candidate media attributes: a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host But when freeswitch makes the INVITE on phone B, these 2 are missing and phone is looking for it, so the INVITE gets rejected by phone with 448 Not acceptable here So the question is, how can I make the freeswitch to pass these candidate media attributes? On Wed, Oct 21, 2009 at 8:58 AM, Jason White ja...@jasonjgw.net wrote: I suspect the codec negotiation. Make sure that both ends are offering a common codec. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-21 17:48 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TCFS = 10.4.62.7
TCSIP phone = 10.4.62.89
TCH323 endpoint = 10.1.14.153
TC
TC
TC
TC TC2. hangup from sip side doesn't release the h323 leg (now the difference
TC is
TC TCthat FS is not complaining about thread mismatch ant it looks clean but
TC FS
TC TCdoesn't send any releasecomplete message... strange)
TC TC3. coredumps when i place outgoing calls
TC
TC btw,
TC
TC TC 70.552157 10.1.14.153 - 10.4.62.7H.245 endSessionCommand
TC TC 70.552401 10.1.14.153 - 10.4.62.7H.225.0 CS: releaseComplete
TC TC 70.55398110.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
TC TC 70.55448810.4.62.7 - 10.1.14.153 H.245 endSessionCommand
TC TC 70.55505610.4.62.7 - 10.1.14.153 H.225.0 CS: releaseComplete
TC
TC it send, now i have no way to test h323-sip transit, i will have it
TC tomorow.
TC sip-h323 for me work fine now, give backtrace from code dump of 1.0.4
TC where it die?
TC
TC
TC
TCthis endSession is when i hangup from H232 side as well :P ... if i don't
TChangup on H323 side the H323 leg is not released. Pls chec the time the
TCpackets were sent ...
TC
TC
TC
TCHere i hangup on the SIP Phone:
TC
TC68.374916 10.4.62.89 - 10.4.62.7SIP Request: BYE
TCsip:mod_so...@10.4.62.7:5060
TC68.37534210.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
TC68.37562010.4.62.7 - 10.4.62.89 SIP Status: 200 OK
TC
TC2 sec delay
TC
TCHere i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
TCfirst here )
TC70.552157 10.1.14.153 - 10.4.62.7H.245 endSessionCommand
TC70.552401 10.1.14.153 - 10.4.62.7H.225.0 CS: releaseComplete
TC
TC
TCFS just acknowlages it here:
TC70.55448810.4.62.7 - 10.1.14.153 H.245 endSessionCommand
TC70.55505610.4.62.7 - 10.1.14.153 H.225.0 CS: releaseComplete
TC
TC
TC
TCI have enabled crash-protection and when i do SIP = H323 call it doesn't
TCgenerate coredumps... it is just this thread that is crashing ... pls check
TCthe log downbelow:
core dump in case enabled crash-protection may be unusable, i have a case then
my module crash silently, after this crash-protection is killing sip leg and
after
this i get core dump where backtrace show me segfault in libc6, i spent one day
to
understand this situation, and then i disable crash-protection i see there is
actualy
it crashes. disable crash-protection and show backtrace of crash, i think
result will
be different.
TC
TCDialplan: sofia/internal/1...@singtel Regex (FAIL) [EMERGENCY]
TCdestination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
TCDialplan: sofia/internal/1...@singtel parsing [default-SPECIAL_SERVICES]
TCcontinue=false
TCDialplan: sofia/internal/1...@singtel Regex (FAIL) [SPECIAL_SERVICES]
TCdestination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
TCDialplan: sofia/internal/1...@singtel parsing [default-ENYTHING_ELSE]
TCcontinue=false
TCDialplan: sofia/internal/1...@singtel Regex (PASS) [ENYTHING_ELSE]
TCdestination_number(05492122) =~
TC/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
TCDialplan: sofia/internal/1...@singtel Action
TCset(effective_caller_id_number=1001282122)
TCDialplan: sofia/internal/1...@singtel Action set(NCX_IP=10.4.4.254)
TCDialplan: sofia/internal/1...@singtel Action set(call_timeout=30)
TCDialplan: sofia/internal/1...@singtel Action set(hangup_after_bridge=true)
TCDialplan: sofia/internal/1...@singtel Action set(bypass_media=false)
TCDialplan: sofia/internal/1...@singtel Action set(proxy_media=true)
TCDialplan: sofia/internal/1...@singtel Action bridge(h323/05492...@${ncx_ip})
TC2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
TC(sofia/internal/1...@singtel) State Change CS_ROUTING - CS_EXECUTE
TC2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
TCsofia/internal/1...@singtel [BREAK]
TC2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
TC(sofia/internal/1...@singtel) State ROUTING going to sleep
TC2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
TC(sofia/internal/1...@singtel) Running State Change CS_EXECUTE
TC2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
TC(sofia/internal/1...@singtel) State EXECUTE
TC2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
TCsofia/internal/1...@singtel SOFIA EXECUTE
TC2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
TCsofia/internal/1...@singtel Standard EXECUTE
TCEXECUTE sofia/internal/1...@singtel set(open=true)
TC2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
TCsofia/internal/1...@singtel SET [open]=[true]
TCEXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
TCEXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
TCEXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
TCEXECUTE sofia/internal/1...@singtelset(effective_caller_id_number=1001282122)
Re: [Freeswitch-users] Qustion about INFO messages after Connect/Answer
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Mike,
just updated my prod system. The 1/a/ problem is solved with Anthony's
originate_callee_id_name chvar.
thanks alot :)
So, last thing of this thread is still the unknown thing on callee's
display, which is (by now) NOT affected by the new chvars.
regards
Helmut
On 19.10.2009 23:35, Michael Collins wrote:
On Mon, Oct 19, 2009 at 1:07 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
please update and test trunk
1) I changed the core to remove the excess data by default in your
scenario
2) I added variables you can use to control it
origination_callee_id_name origination_callee_id_number which belong
in {} in the dial string eg
{origination_callee_id_number=1234}openzap/1/a/1234
After you test, please confirm the behavior and then we'll update the
wiki on these two new chan vars.
-MC
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Version: GnuPG v1.4.7 (MingW32)
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Alf3NeoCA8Qbm2PZ1k2HHOg=
=hzVn
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Re: [Freeswitch-users] NOT in dialplan expression
On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker r...@rupa.com wrote: You can also look at using anti-action rather than action after the condition. condition == if action == then anti-action == else Rupa Leon, Nice job of explaining the options. Sometimes we forget about the powerful constructs that are available in FreeSWITCH and PCRE. karma++ for both of you. :) -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] Qustion about INFO messages after Connect/Answer
On Wed, Oct 21, 2009 at 10:43 AM, Helmut Kuper helmut.ku...@ewetel.dewrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Mike, just updated my prod system. The 1/a/ problem is solved with Anthony's originate_callee_id_name chvar. thanks alot :) So, last thing of this thread is still the unknown thing on callee's display, which is (by now) NOT affected by the new chvars. Okay, you are able to reproduce that unknown thing? Can you pastebin a fresh debug log w/ SIP trace on, plus and relevant dp changes from the default dialplan? Thanks, MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] 3rd Party Dial Plan Tool
Can anyone recommend a good 3rd party dialplan tool that will work with Freeswitch? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Proxy media mode with T.38 re-invite
can you try trunk and let me know right away, if it's still not working i may need ssh access and call you on the phone. On Tue, Oct 20, 2009 at 10:50 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Done. On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale anthony.miness...@gmail.com wrote: issue: console loglevel debug sofia profile internal siptrace on and put it on pastebin http://pastebin.freeswitch.org -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] 3rd Party Dial Plan Tool
none specifically exist... good ole trusty editor? /b PS: http://www.cudatel.com On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote: Can anyone recommend a good 3rd party dialplan tool that will work with Freeswitch? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] 3rd Party Dial Plan Tool
You could also try consult...@freeswitch.org On Wed, Oct 21, 2009 at 14:00, Brian West br...@freeswitch.org wrote: none specifically exist... good ole trusty editor? /b PS: http://www.cudatel.com On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote: Can anyone recommend a good 3rd party dialplan tool that will work with Freeswitch? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Shannon ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sched_api doesn't get launched
The syntax is different, but the api is the same as lua: So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) create the API object and use the execute method of it. Mike On Oct 21, 2009, at 5:44 AM, Henry Huang wrote: I can't seem to find the right thing to use in mod_java to execute api commands, only api_after_bridge 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1688...@192.168.1.66! # # A fatal error has been detected by the Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624 # # JRE version: 6.0_16-b01 # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 ) # Problematic frame: # C [libc.so.6+0x6f480] strcpy+0x10 # # An error report file with more information is saved as: # /usr/local/freeswitch/bin/hs_err_pid1927.log 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid Application sched_api 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup sofia/internal/1688...@192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # The crash happened outside the Java Virtual Machine in native code. # See problematic frame for where to report the bug. On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.org wrote: On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang red.rain.se...@gmail.com wrote: So how would you trigger it from a script dialplan? The only time it seemed to work is when I did setVariable(api_after_bridge, sched_api blah blah blah); but then it gets executed after the channel's been teared down. I thought api_after_bridge means right after the call gets connected. I need something to execute an api command right before or right after the call gets bridged. api_after_bridge is a channel variable, so using setVariable works just fine. If you need to sched_api is an API only. Check these out: http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) Remember, if the method you are using isn't found in the dial plan tools then it isn't a dial plan application. Make sure it's on the list: http://wiki.freeswitch.org/wiki/Mod_dptools On the other hand, API commands are listed here: http://wiki.freeswitch.org/wiki/Mod_commands dptools require a session object, api commands require an api object... -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
Yes, I think you are right, the sip stack comes from company called M5T as far as I know, and there are references to ICE on their site. On Wed, Oct 21, 2009 at 7:29 PM, Michael Jerris m...@jerris.com wrote: This appears to be some sort of ice implementation? We don't support sip ice at this time. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sched_api doesn't get launched
Yes you need an API object as described in other email. Which line of code from java caused that segfault It looks like a simple NULL string issue that we may want to hunt down. On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang red.rain.se...@gmail.comwrote: I can't seem to find the right thing to use in mod_java to execute api commands, only api_after_bridge 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1688...@192.168.1.66! # # A fatal error has been detected by the Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624 # # JRE version: 6.0_16-b01 # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 ) # Problematic frame: # C [libc.so.6+0x6f480] strcpy+0x10 # # An error report file with more information is saved as: # /usr/local/freeswitch/bin/hs_err_pid1927.log 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid Application sched_api 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup sofia/internal/1688...@192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # The crash happened outside the Java Virtual Machine in native code. # See problematic frame for where to report the bug. On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.orgwrote: On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang red.rain.se...@gmail.comwrote: So how would you trigger it from a script dialplan? The only time it seemed to work is when I did setVariable(api_after_bridge, sched_api blah blah blah); but then it gets executed after the channel's been teared down. I thought api_after_bridge means right after the call gets connected. I need something to execute an api command right before or right after the call gets bridged. api_after_bridge is a channel variable, so using setVariable works just fine. If you need to sched_api is an API only. Check these out: http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) Remember, if the method you are using isn't found in the dial plan tools then it isn't a dial plan application. Make sure it's on the list: http://wiki.freeswitch.org/wiki/Mod_dptools On the other hand, API commands are listed here: http://wiki.freeswitch.org/wiki/Mod_commands dptools require a session object, api commands require an api object... -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- 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] Calls dropping on SIP timer expiry due to UPDATE's being ignored.
This should now be fixed in latest svn trunk. Mike On Oct 21, 2009, at 12:45 PM, Keith Laaks wrote: Hi, Hope someone knows how I am able to get around this one. Here goes... Did an upgrade to trunk (from a July vintage build) last week and noticed calls out to a provider were now failing after about 30 seconds or so - post answer. Tried latest (15183) - same thing. Analysing, I see that I have multiple UPDATE messages now being sent to the provider, but no response being sent back to FS. So FS times out and eventually kills the call. Interestingly, it only drops the A-leg; the B-leg remains up till the B party hangs up. I cant recall seeing these UPDATE messages before... The intent of the UPDATE seems to be to send the callee name number to the B-leg. If its the provider's sip stack that's broken w.r.t. handling UPDATE - is there any way to get around it by doing something in my config to ensure these UPDATE's are not 'triggered' ? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] NOT in dialplan expression
On Wed, Oct 21, 2009 at 3:51 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Thanks Guys for your suggestions! Very much appreciated.
The reason I asked was that I want to check if a variable is set or
not. I don't think your suggestions will work because I actually have
to check to see an undefined variable. Any way to do this?
Not in the dialplan itself. Better off using a script for this.
I want to have the same extension for checking digits read from DTMF.
The problem is that the digits are not set to the variable until the
transfer statement, which means I would require 2 extensions. It
will make things simpler if I can have the same extension that reads
the digits and then checks them.
You need either to use multiple extensions or use a script. The dialplan was
specifically designed not to be a programming language and the functionality
you seek is best served by using a script or by using transfer to drop the
call through the diaplan again and use another extension.
-MC
For example:
extension name=checkdigits
condition field=destination_number expression=^checkdigits$/
!--The next condition should be true if the variable $digits
is not set/undefined --
condition field=${digits} expression=^[^.+]$ /
- what should be here to check for undefined variable $digits
?
action application=read data=1 10 ivr/ivr-hello.wav
digits 1 #/
action application=phrase data=spell,${digits}/
action application=transfer data=checkdigits digits XML
features/
/condition
condition field=${pincode} expression=^${some_variable}$
SOME ACTIONS HERE
/condition
/extension
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Re: [Freeswitch-users] NOT in dialplan expression
Like rupa said,
use anti-action?
Remember FreeSWITCH XML dialplan is not a programming language it's a
pre-processor routing markup
Se potential use of anti-action below:
extension name=checkdigits
condition field=destination_number expression=^checkdigits$/
!--The next condition should be true if the variable $digits is not
set/undefined --
condition field=${digits} expression=^[^.+]$ /
--action application=read data=1 10 ivr/ivr-hello.wav digits
1 #/
action application=phrase data=spell,${digits}/
action application=transfer data=checkdigits digits XML
features/
anti-action application=set data=not_true=true/
/condition
condition field=${pincode} expression=^${some_variable}$
SOME ACTIONS HERE
/condition
/extension
On Wed, Oct 21, 2009 at 5:51 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Thanks Guys for your suggestions! Very much appreciated.
The reason I asked was that I want to check if a variable is set or
not. I don't think your suggestions will work because I actually have
to check to see an undefined variable. Any way to do this?
I want to have the same extension for checking digits read from DTMF.
The problem is that the digits are not set to the variable until the
transfer statement, which means I would require 2 extensions. It
will make things simpler if I can have the same extension that reads
the digits and then checks them.
For example:
extension name=checkdigits
condition field=destination_number expression=^checkdigits$/
!--The next condition should be true if the variable $digits
is not set/undefined --
condition field=${digits} expression=^[^.+]$ /
- what should be here to check for undefined variable $digits
?
action application=read data=1 10 ivr/ivr-hello.wav
digits 1 #/
action application=phrase data=spell,${digits}/
action application=transfer data=checkdigits digits XML
features/
/condition
condition field=${pincode} expression=^${some_variable}$
SOME ACTIONS HERE
/condition
/extension
On Thu, Oct 22, 2009 at 4:44 AM, Michael Collins m...@freeswitch.org
wrote:
On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker r...@rupa.com wrote:
You can also look at using anti-action rather than action after the
condition.
condition == if
action == then
anti-action == else
Rupa Leon,
Nice job of explaining the options. Sometimes we forget about the
powerful
constructs that are available in FreeSWITCH and PCRE. karma++ for both of
you. :)
-MC
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--
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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Re: [Freeswitch-users] UUID of the newly originated call?
which revision of FS are you using? On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. nagale...@gmail.com wrote: I've tried with origination_uuid. First, I tried with SIP and my program executes successfully as what I expected. This program initiates a new call when a call comes and let the new call to eavesdrop the landed call. When, I experimented with PRI(openzap), I'm facing the following error. And, I am unable to make any calls(even from CLI). It is reporting the same error for the subsequent calls. Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9] Full log is in http://pastebin.freeswitch.org/10780 My script is here, http://pastebin.freeswitch.org/10781 What is this error for and how to avoid this? Is there any other way to get the uuid of the originated call except explicitly defining(origination_uuid).?! -- Regards, Nagalenoj H. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] CS_REPORTING Channel event state
better still, 1) update to trunk with make current 2) Try to reproduce, if you can: a) open a new shell b) make sure you have gdb and gcore installed c) run ./support-d/fscore_pb gcore d) report the URL On Tue, Oct 20, 2009 at 8:48 PM, Michael Jerris m...@jerris.com wrote: REPORTING is the state that it writes to CDR. If you have calls stuck in this state, take one and try to use uuid_kill on it and see if it goes away, then get a core off of it and pastebin the thread apply all bt (with no other calls up). What modules are you using for cdr and with what configuration? Mike On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote: Dear All What's CS_REPORTING state ? I found many channels not hang up ans state is CS_REPORTING e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20 21:50:26,1256050226,sofia/external/x...@xxx.xxx.xx. 191:7050,CS_REPORTING,FreeSWITCH,,xx.xxx.xxx.xxx, 7050,,,XML,public, ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH-users Digest, Vol 40, Issue 179
Hi,
Thanks for reply, it really helped me. One more thing to ask, how can we
make decision against ,, =, = in condition header? Like we use == for
action and != for anti-action.
Kindly highlight it.
-- Forwarded message --
From: Ahmed Munir ahmedmunir...@gmail.com
To: FreeSwitch freeswitch-users@lists.freeswitch.org
Date: Wed, 21 Oct 2009 15:37:15 +0500
Subject: [Freeswitch-users] Call custom variable in condition
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
-- Forwarded message --
From: Ghulam Mustafa mustafa...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 21 Oct 2009 15:52:24 +0500
Subject: Re: [Freeswitch-users] Call custom variable in condition
Ahmed,
you can't use variables set by set application within a condition, though
it doesn't make sense. wondering if there is any logic behind this or it's
just a simple missing feature. anyone?
-m
Ahmed Munir wrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
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-- Forwarded message --
From: Tihomir Culjaga tculj...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 21 Oct 2009 13:13:13 +0200
Subject: Re: [Freeswitch-users] Call custom variable in condition
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=${AUTHENTICATION_STATUS} expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
here is one of my dialplan. I'm using execute_extension but it is quite the
same...
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
Can't you use the inline statement to set a variable so that it can be
used directly in a condition?
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
On Thu, Oct 22, 2009 at 3:08 PM, Ahmed Munir ahmedmunir...@gmail.com wrote:
Hi,
Thanks for reply, it really helped me. One more thing to ask, how can we
make decision against ,, =, = in condition header? Like we use == for
action and != for anti-action.
Kindly highlight it.
-- Forwarded message --
From: Ahmed Munir ahmedmunir...@gmail.com
To: FreeSwitch freeswitch-users@lists.freeswitch.org
Date: Wed, 21 Oct 2009 15:37:15 +0500
Subject: [Freeswitch-users] Call custom variable in condition
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
-- Forwarded message --
From: Ghulam Mustafa mustafa...@gmail.com
To: freeswitch-us...@lists.freeswitch.org
Date: Wed, 21 Oct 2009 15:52:24 +0500
Subject: Re: [Freeswitch-users] Call custom variable in condition
Ahmed,
you can't use variables set by set application within a condition,
though it doesn't make sense. wondering if there is any logic behind this or
it's just a simple missing feature. anyone?
-m
Ahmed Munir wrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do
implement it(Note: I don't want to call script). And one more thing to ask
how can I transfer the values within the same context?
--
Regards,
Ahmed Munir
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-- Forwarded message --
From: Tihomir Culjaga tculj...@gmail.com
To: freeswitch-us...@lists.freeswitch.org
Date: Wed, 21 Oct 2009 13:13:13 +0200
Subject: Re: [Freeswitch-users] Call custom variable in condition
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=${AUTHENTICATION_STATUS} expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
here is one of my dialplan. I'm using execute_extension but it is quite
the same...
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum
Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
On Wed, Oct 21, 2009 at 9:28 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Can't you use the inline statement to set a variable so that it can be used directly in a condition? http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions Yes you can. Just don't abuse it like Tony said. If you find yourself doing it a lot then it's a sure sign that you're taking the wrong approach. -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] FreeSWITCH-users Digest, Vol 40, Issue 179
On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir ahmedmunir...@gmail.comwrote: Hi, Thanks for reply, it really helped me. One more thing to ask, how can we make decision against ,, =, = in condition header? Like we use == for action and != for anti-action. Kindly highlight it. You can only do greater than and less than in the date/time matching. See the date/time example in the default.xml dialplan file. You can also use regular expressions if you're in a pinch. For example, if you need to match numbers = 1100 and = 1500 you could just use this regex: ^(1[1234]\d\d|1500)$ The real question, though, is this: what types of values do you need to match for GT or LT? Date/time? Money? Other? That will determine if you need to use a script or just the dialplan. -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
