[Freeswitch-users] send sip options message
hi does anybody know how to send a sip options message to a registered user, using the event socket or something else build in freeswitch i think the ping parameter does something like this for gateways. what i want/need is the same thing that is provided in asterisk with the qualifying option, to see how "reachable" a certain client is. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] conference question
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is started with 323963...@conf+flags{mute,waste}
waste is needed to get the whole audio stream
it now happens that listeners sometimes hear each other, that shouldn't be
what can i do to resolve this problem?
we are using version 1.0.4
br
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Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP
xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
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E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] conference question
thank you for your response
as a listener waste influences what you hear and mute say's you cannot speak
this is what our customer wanted because the speaker is the only one who is
heard in this "conference" or meeting room - this rooms are for lectures
we tried to disable waste for the listeners (and let it on for the speaker) but
this
resulted in "choppy" sound for the listeners (silence periods between words and
sentences)
i hope i could explain my problem a little bit better
br
On 2009-09-01 23:32, Bradley Brashier wrote:
> I haven't really used waste much myself, but my understanding is that
> waste and mute would conflict, since waste says "send audio always"
> and mute says "send audio never". I didn't understand why you're using
> waste on the listeners... you should be able to get by with waste just
> on the speaker (again, that's how I understand it).
>
> 2009/9/1 Christian Löschenkohl:
>> hello
>>
>> we have got a little problem with the conference application
>> in our setup we have da system for customers where speakers can dial in
>> with phonenumber+1 and the listeners dial in with phonenumber
>>
>> the speakers conference is started with 323963...@conf+flags{waste}
>> the listeners conference is started with 323963...@conf+flags{mute,waste}
>>
>> waste is needed to get the whole audio stream
>> it now happens that listeners sometimes hear each other, that shouldn't be
>>
>> what can i do to resolve this problem?
>> we are using version 1.0.4
>>
>> br
>>
>> --
>> Ing. Christian Löschenkohl
>> Technische Leitung, Forschung& Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation& Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T +43 (0) 5 77 11 - 1000
>> F +43 (0) 5 77 11 - 1002
>> E christian.loeschenk...@xpirio.com
>>
>> ___
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>
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Technische Leitung, Forschung & Entwicklung VoIP
xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
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E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] conference question
thank you but we defined the conference with so no keys should be available br On 2009-09-02 17:02, Andy Spitzer wrote: > Woof! > > On Tue, 01 Sep 2009 18:52:01 -0400, Anthony > Minessale wrote: > >> there is no chance that you would not enter the conf muted the way you >> describe unless you are using an older revision of FS that had a bug in >> the parsing of the conference flags. > > Perhaps some listeners are hitting the "unmute" DTMF key? > > --Woof! > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] stability problems
hello we have regular (every 4-6 days) stability problems with freeswitch when the problme occurs - no registers are done bythe server (olny 1 ack of the initial register) - no more calls are working - the calls are all ending with a timeout (cdr caues ORIGINATOR_CANCEL) - only a restart of the whole server cures the problem the server doesn't crash or segfault my first try was to enable the crash-protection flag, but with no difference the server is restartet every night and the last stand still was after about 15h uptime the system is an sun fire 2400 with debian 64 bit system how could i offer you more information to solve this big problem br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] stability problems
on debian lenny amd64 with the build-essential package an then with ./configure --prefix=/opt/freeswitch make make install nothing else br On 2009-09-03 16:12, Brian West wrote: > Sounds like you have some build skew... can you tell us how you built > FreeSWITCH? > > /b > > On Sep 3, 2009, at 2:29 AM, Christian Löschenkohl wrote: > >> hello >> >> we have regular (every 4-6 days) stability problems with freeswitch >> when the problme occurs >> >> - no registers are done bythe server (olny 1 ack of the initial >> register) >> - no more calls are working >> - the calls are all ending with a timeout (cdr caues >> ORIGINATOR_CANCEL) >> - only a restart of the whole server cures the problem >> >> the server doesn't crash or segfault >> my first try was to enable the crash-protection flag, but with no >> difference >> the server is restartet every night and the last stand still was >> after about 15h uptime >> >> the system is an sun fire 2400 with debian 64 bit system >> >> how could i offer you more information to solve this big problem >> >> br >> >> -- >> Ing. Christian Löschenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenk...@xpirio.com >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] stability problems
sorry i can not follow you i build everthing from scratch (download source, unpack and build) what i mean with the build-essential package is a debian meta package that contains gcc, make and so on br On 2009-09-03 18:25, Brian West wrote: > Can you try NOT using a package? I have a theory that the package has > a few optimization flags in it that breaks things. > > /b > > On Sep 3, 2009, at 10:56 AM, Rupa Schomaker wrote: > >> Since you are using the debian package, the files will be in >> /opt/freeswitch not /usr/local/freeswitch. > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] stability problems
yes, thank you for this i will follow your instructions an remove any files from older builds and reinstall - then I'll give feedback hope this resolves my problem br On 2009-09-03 19:11, Anthony Minessale wrote: > I already provided exact instructions. > > > On Thu, Sep 3, 2009 at 11:50 AM, Brian West <mailto:br...@freeswitch.org>> wrote: > > Please join IRC if you experience the issue again #freeswitch on > irc.freenode.net <http://irc.freenode.net> > > /b > > > On Sep 3, 2009, at 11:43 AM, Christian Löschenkohl wrote: > > > sorry i can not follow you > > i build everthing from scratch (download source, unpack and build) > > > > what i mean with the build-essential package is a debian meta package > > that contains gcc, make and so on > > > > br > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > 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 > <mailto:msn%3aanthony_miness...@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > <mailto:paypal%3aanthony.miness...@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > <mailto:sip%3a...@conference.freeswitch.org> > iax:gu...@conference.freeswitch.org/888 > <http://iax:gu...@conference.freeswitch.org/888> > googletalk:conf+...@conference.freeswitch.org > <mailto:googletalk%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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] restart when convenient
hello i'm looking for a possibility to restart freeswitch like it is possible with asterisk. for me i tried to created a script that looks for open channels and if no channel is open it restarts freeswitch with the init script (not the most efficient way). i think i would be great if we would have a buildin function for this, i think such command would help with maintenance and not only for me. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] patch for debian init script
hi
just a quick patch for the debian init script debian/freeswitch.init
i do use the reload function and the script complains about the -C option
it also would be perfect if the reload option is enabled by default (usefull
for logrotating) -> combined in second patch
br
*
--- debian/freeswitch.init-old 2009-09-08 16:18:49.0 +0200
+++ debian/freeswitch.init 2009-09-08 16:19:13.0 +0200
@@ -103,7 +103,7 @@
# restarting (for example, when it is sent a SIGHUP),
# then implement that here.
#
- start-stop-daemon -C $USER --stop --signal 1 --quiet --pidfile $PIDFILE
--name $NAME
+ start-stop-daemon -c $USER --stop --signal 1 --quiet --pidfile $PIDFILE
--name $NAME
return 0
}
*
--- debian/freeswitch.init-old 2009-09-08 16:26:01.0 +0200
+++ debian/freeswitch.init 2009-09-08 16:19:13.0 +0200
@@ -103,7 +103,7 @@
# restarting (for example, when it is sent a SIGHUP),
# then implement that here.
#
- start-stop-daemon -C $USER --stop --signal 1 --quiet --pidfile $PIDFILE
--name $NAME
+ start-stop-daemon -c $USER --stop --signal 1 --quiet --pidfile $PIDFILE
--name $NAME
return 0
}
@@ -124,15 +124,15 @@
2) [ "$VERBOSE" != no ] && log_end_msg 1 ;;
esac
;;
- reload|force-reload)
+ #reload|force-reload)
#
# If do_reload() is not implemented then leave this commented out
# and leave 'force-reload' as an alias for 'restart'.
#
- log_daemon_msg "Reloading $DESC" "$NAME"
- do_reload
- log_end_msg $?
- ;;
+ #log_daemon_msg "Reloading $DESC" "$NAME"
+ #do_reload
+ #log_end_msg $?
+ #;;
restart|force-reload)
#
# If the "reload" option is implemented then remove the
@@ -156,8 +156,8 @@
esac
;;
*)
- echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2
- #echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2
+ #echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2
+ echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2
exit 3
;;
esac
--
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP
xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] patch for debian init script
is done
br
On 2009-09-08 18:26, Michael Jerris wrote:
> Please post this patch to http://jira.freeswitch.org in the build
> system project and I will get this merged in
>
> Mike
>
> On Sep 8, 2009, at 10:28 AM, Christian Löschenkohl wrote:
>
>> hi
>>
>> just a quick patch for the debian init script debian/freeswitch.init
>> i do use the reload function and the script complains about the -C
>> option
>> it also would be perfect if the reload option is enabled by default
>> (usefull
>> for logrotating) -> combined in second patch
>>
>> br
>>
>> *
>>
>> --- debian/freeswitch.init-old 2009-09-08 16:18:49.0 +0200
>> +++ debian/freeswitch.init 2009-09-08 16:19:13.0 +0200
>> @@ -103,7 +103,7 @@
>> # restarting (for example, when it is sent a SIGHUP),
>> # then implement that here.
>> #
>> - start-stop-daemon -C $USER --stop --signal 1 --quiet --
>> pidfile $PIDFILE --name $NAME
>> + start-stop-daemon -c $USER --stop --signal 1 --quiet --
>> pidfile $PIDFILE --name $NAME
>> return 0
>> }
>>
>> *
>>
>> --- debian/freeswitch.init-old 2009-09-08 16:26:01.0 +0200
>> +++ debian/freeswitch.init 2009-09-08 16:19:13.0 +0200
>> @@ -103,7 +103,7 @@
>> # restarting (for example, when it is sent a SIGHUP),
>> # then implement that here.
>> #
>> - start-stop-daemon -C $USER --stop --signal 1 --quiet --
>> pidfile $PIDFILE --name $NAME
>> + start-stop-daemon -c $USER --stop --signal 1 --quiet --
>> pidfile $PIDFILE --name $NAME
>> return 0
>> }
>>
>> @@ -124,15 +124,15 @@
>> 2) [ "$VERBOSE" != no ]&& log_end_msg 1 ;;
>> esac
>> ;;
>> - reload|force-reload)
>> + #reload|force-reload)
>> #
>> # If do_reload() is not implemented then leave this
>> commented out
>> # and leave 'force-reload' as an alias for 'restart'.
>> #
>> - log_daemon_msg "Reloading $DESC" "$NAME"
>> - do_reload
>> - log_end_msg $?
>> - ;;
>> + #log_daemon_msg "Reloading $DESC" "$NAME"
>> + #do_reload
>> + #log_end_msg $?
>> + #;;
>> restart|force-reload)
>> #
>> # If the "reload" option is implemented then remove the
>> @@ -156,8 +156,8 @@
>> esac
>> ;;
>> *)
>> - echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-
>> reload}">&2
>> - #echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}"
>>> &2
>> + #echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-
>> reload}">&2
>> + echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}"
>>> &2
>> exit 3
>> ;;
>> esac
>>
>>
>> --
>> Ing. Christian Löschenkohl
>> Technische Leitung, Forschung& Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation& Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T +43 (0) 5 77 11 - 1000
>> F +43 (0) 5 77 11 - 1002
>> E christian.loeschenk...@xpirio.com
>
>
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--
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP
xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] stability problems
hello anthony i'm sorry the cleanup didn't solve my problem i have opend a jira bug n this - key FSCORE-432 hope this is right br On 2009-09-03 17:02, Anthony Minessale wrote: > Which revision are you using? > > If you are not running the latest trunk, please upgrade to that in case > your problem requires us to change the code > we need it to be up to date. > > > 1) Remove any binary files which may get mixed in from an older build > rm /usr/local/freeswitch/bin/* > rm /usr/local/freeswitch/lib/* > rm /usr/local/freeswitch/mod > 2) Build Latest Trunk > 3) Reproduce the problem. > > If you get the problem keep FreeSWITCH running and capture a gcore back > trace. > > ./scripts/freeswitch-gcore > gcore.txt > > Send us the file as an attachment or attached to a new jira issue. > http://jira.freeswitch.org > > > > > > > 2009/9/3 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > on debian lenny amd64 with the build-essential package > > an then with > > ./configure --prefix=/opt/freeswitch > make > make install > > nothing else > > br > > On 2009-09-03 16:12, Brian West wrote: > > Sounds like you have some build skew... can you tell us how you built > > FreeSWITCH? > > > > /b > > > > On Sep 3, 2009, at 2:29 AM, Christian Löschenkohl wrote: > > > >> hello > >> > >> we have regular (every 4-6 days) stability problems with freeswitch > >> when the problme occurs > >> > >> - no registers are done bythe server (olny 1 ack of the initial > >> register) > >> - no more calls are working > >> - the calls are all ending with a timeout (cdr caues > >> ORIGINATOR_CANCEL) > >> - only a restart of the whole server cures the problem > >> > >> the server doesn't crash or segfault > >> my first try was to enable the crash-protection flag, but with no > >> difference > >> the server is restartet every night and the last stand still was > >> after about 15h uptime > >> > >> the system is an sun fire 2400 with debian 64 bit system > >> > >> how could i offer you more information to solve this big problem > >> > >> br > >> > >> -- > >> Ing. Christian Löschenkohl > >> Technische Leitung, Forschung& Entwicklung VoIP > >> > >> xpirio > >> Telekommunikation& Service GmbH > >> Lakeside B04 > >> 9020 Klagenfurt > >> Austria > >> > >> T +43 (0) 5 77 11 - 1000 > >> F +43 (0) 5 77 11 - 1002 > >> E christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > >> > >> ___ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > > > ___ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian Löschenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://w
[Freeswitch-users] memory leak - outbound socket
hello version : 1.0.4 std. tarball - the wiki example for php outbound socket connection leaks memory without the async option - the memory used is never given back - async isn't that usefull for us - we want to query databases, set variables and so on no wait statements are possible <<<< no async the script is on the site http://wiki.freeswitch.org/wiki/PHP_ESL --- what can i do? on our production server we use outbound socket connection and the 4 gig of memory are eaten up in less than a day br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] memory leak - outbound socket
as a good fs user - of course i am :-) - i made a jira on this MODAPP-336 to be precise i hope this helps to solve my problem br On 2009-09-16 17:05, Rupa Schomaker wrote: > Either: > > 1) Provide a simple self-contained example that demonstrates the leak > > or > > 2) Run your application with FreeSWITCH under valgrind and provide the > final output. To run freeswitch under valgrind: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29 > > You should not have to run with high load to capture the behavior. > Try with just 5 (in series) and then stop freeswitch. > > > 2009/9/16 Christian Löschenkohl: >> hello >> >> version : 1.0.4 std. tarball >> >> - the wiki example for php outbound socket connection leaks memory without >> the async option >> - the memory used is never given back >> - async isn't that usefull for us - we want to query databases, set >> variables and so on >>no wait statements are possible >> >> >> >> > data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/> >> <<<< no async >> >> >> >> >> the script is on the site >> http://wiki.freeswitch.org/wiki/PHP_ESL >> >> --- >> >> what can i do? >> on our production server we use outbound socket connection and the 4 gig of >> memory are >> eaten up in less than a day >> >> br >> >> -- >> Ing. Christian Löschenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenk...@xpirio.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 >> > > > -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] local stream problem - moh
hello since installing the latest trunk 14894 my local streams / moh don't work anymore no config file has changed, the files are in place show_local_stream outputs default,/opt/freeswitch/sounds/music/8000 moh/16000,/opt/freeswitch/sounds/music/16000 moh/8000,/opt/freeswitch/sounds/music/8000 console prints 2009-09-17 01:15:01.496277 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.516281 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.516281 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.536282 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.536282 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.556286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.556286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.576286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.576286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.596283 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.596283 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.616261 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.616261 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.636304 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.636304 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.656273 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.656273 [ERR] switch_core_file.c:152 File [(null)] not created! - and yes, files are there ls /opt/freeswitch/sounds/music/8000 danza-espanola-op-37-h-142-xii-arabesca.wav partita-no-3-in-e-major-bwv-1006-1-preludio.wav ponce-preludio-in-e-major.wav suite-espanola-op-47-leyenda.wav - br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] check to see if freeswitch is alive
hello we have the problem here that our freeswitch server "freezes" from time to time (no sip traffic is possible any more). has somebody here expirience or any idea how to monitor freeswitch to answer the questions - is the server alive - does he process sip messages (invites, registrations) and so on the monitoring should work via cron local and remote (with esl for example) a simple "fs_cli -x 'version'" wouldn't work - it's too simple, am i right? any ideas would be helpful br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] check to see if freeswitch is alive
i know, we tried to to get closer to this with anthony (see thread "stability problems", started on 03.09.2009). the old issue described a server that used more profiles (8-10) and some event socket scripts. we came to the conclusion that it could be a race condition when using multiple profiles. now on the server we need monitor we use only internal functions and conditions - no external scripts - no database - only on profile - only freeswitch stock functions the symptoms are the same - no more progress displays on the fs_cli console (debug level info) - invites are answered with trying, but no more communication after this - customers said that option requests are answered (they could not switch to fallback because this worked) we did use the trunk version and have now updated to 15162, but i don't think our problem is solved with any update so far. br On 2009-10-19 18:39, Michael Collins wrote: > > > 2009/10/19 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > hello > > we have the problem here that our freeswitch server "freezes" from time > to time (no sip traffic is possible any more). > > > Monitoring is definitely important, but I'm sure the FS devs would like > to know more about what happens when FS "freezes" - is it FS or some > other process that is messing things up? If there's an issue with FS > then the developers would like to know more. > -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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] check to see if freeswitch is alive
thank you for the script, we will try this asap br On 2009-10-19 18:28, Andy Spitzer wrote: > Woof! > > On Mon, 19 Oct 2009 12:07:10 -0400, Christian Löschenkohl > wrote: > >> any ideas would be helpful > > We run a perl script that checks if the servers are responding to > requests. It can send OPTIONS, and PING requests to various servers > periodically. If the response it gets back isn't correct, it can send > e-mail. It'll keep testing and send an e-mail again when the system is > working again. It's defaults setup for testing sipXecs servers, but it's > easy enough to set the various command line arguments to point it at > FreeSWITCH directly. > > Code here. > > http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in > > --Woof! > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to rewrite freeswitch SDP
hi but this wouldn't work for larger volumens, g729 and t.38 or am i wrong on this? br On 2009-11-11 18:23, Kristian Kielhofner wrote: > This might be a bit too obvious but unless you have a specific reason > to use proxy_media (handling goofy codecs is a big one) you could just > set proxy_media=false and FreeSWITH will proxy the media (effectively) > and rewrite the entire SDP by default. > > On Wed, Nov 11, 2009 at 12:08 PM, Juan Backson wrote: >> Hi, >> >> I am using 1.0.4 version of freeswitch and I am doing proxy_media for all >> calls. Basically, I just proxy all media from one gateway to another with >> freeswitch serving as a middleman. >> >> In the outgoing invite, I found that the owner line ( o= ) in SDP is showing >> the originator's IP which I would like to avoid. >> >> Is there anyway to rewirte part of the SDP for the outgoing invite? >> >> thanks, >> jb >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a "100 Trying" and then a "180 Ringing" within the "180 Ringing" we get a sdp with "a=sendonly" then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no "a=sendonly" would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
thank you for your answer the relevant part of the log is 2009-11-23 21:46:49.625130 [NOTICE] sofia.c:3693 Pre-Answer sofia/interconnect/24785214448370...@38.105.229.100! 2009-11-23 21:46:49.625130 [INFO] sofia.c:3706 Sending early media 2009-11-23 21:46:49.625130 [ERR] sofia_glue.c:2029 No audio codec available 2009-11-23 21:46:49.625130 [NOTICE] switch_channel.c:2048 Hangup sofia/interconnect/nob...@81.94.55.100 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] it's the same with g729 and alaw (refering to brian) in my opinion the ringing here should be generated near end and no audio codec has to be used here (180 ringing) br On 2009-11-23 20:45, Anthony Minessale wrote: > you need to provide a FS console trace of your problem > > from your FS source dir (build root) > > cd libs/esl > make perlmod > cd perl > perl logger.pl <http://logger.pl> -pb christian > > reproduce > > > then hit ctl-c and tell me the url it posted to. > > > > 2009/11/23 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a > "180 Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our > freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think > it would work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus > side - this would > also work (i think). in fact he says that 180 ringing is vaild, he > isn't that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian Löschenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > 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 > <mailto:msn%3aanthony_miness...@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > <mailto:paypal%3aanthony.miness...@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > <mailto:sip%3a...@conference.freeswitch.org> > iax:gu...@conference.freeswitch.org/888 > <http://iax:gu...@conference.freeswitch.org/888> > googletalk:conf+...@conference.freeswitch.org > <mailto:googletalk%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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: > Well its also G729 so I suspect you don't have G729 > > /b > > On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: > >> hi >> >> our freeswitch server has to talk to a sonus ip-switch >> when we want to setup a call we do get a "100 Trying" and then a >> "180 Ringing" >> within the "180 Ringing" we get a sdp with "a=sendonly" then our >> freeswitch >> quits with a CANCEL message. >> i simply don't get why our freeswitch aborts the session - i think >> it would work >> if no "a=sendonly" would be present in the sdp. >> >> my technical contact doesn't want to switch 180 to 183 on the sonus >> side - this would >> also work (i think). in fact he says that 180 ringing is vaild, he >> isn't that wrong in >> this case. >> >> our freeswitch works in proxy mode, we do use trunk 15396 >> see a ngrep trace under http://pastebin.freeswitch.org/11235 >> >> 92.63.208.36 - freeswitch >> 38.105.229.100 - sonus >> >> br >> >> -- >> Ing. Christian Löschenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenk...@xpirio.com >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
sorry about wasting your time (wasn't my intent) the log is at http://pastebin.freeswitch.org/11240 i called 5214448370068 (also other calls are in the log) they now have changed 180 to 183 on the sonus, but makes no difference here br On 2009-11-23 22:07, Anthony Minessale wrote: > do you have the ringback variable set on the channel? > if so it will cause 180 to attempt to play inband ringback indication > > I have nothing left to say because I asked for the whole log with the > siptrace enables not just 5 lines of it. > If you still want help, give me the log to examine and I will tell you > what your problem is. > > > > 2009/11/23 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > thany ou for your answer > > we use g729 on all our other connections in passthrough mode and it > also doesn't work with alaw. > so i don't think it's related to this. > > br > > > On 2009-11-23 20:48, Brian West wrote: > > Well its also G729 so I suspect you don't have G729 > > > > /b > > > > On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: > > > >> hi > >> > >> our freeswitch server has to talk to a sonus ip-switch > >> when we want to setup a call we do get a "100 Trying" and then a > >> "180 Ringing" > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > >> freeswitch > >> quits with a CANCEL message. > >> i simply don't get why our freeswitch aborts the session - i think > >> it would work > >> if no "a=sendonly" would be present in the sdp. > >> > >> my technical contact doesn't want to switch 180 to 183 on the sonus > >> side - this would > >> also work (i think). in fact he says that 180 ringing is vaild, he > >> isn't that wrong in > >> this case. > >> > >> our freeswitch works in proxy mode, we do use trunk 15396 > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > >> > >> 92.63.208.36 - freeswitch > >> 38.105.229.100 - sonus > >> > >> br > >> > >> -- > >> Ing. Christian Löschenkohl > >> Technische Leitung, Forschung& Entwicklung VoIP > >> > >> xpirio > >> Telekommunikation& Service GmbH > >> Lakeside B04 > >> 9020 Klagenfurt > >> Austria > >> > >> T +43 (0) 5 77 11 - 1000 > >> F +43 (0) 5 77 11 - 1002 > >> E christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > >> > >> ___ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > > > ___ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian Löschenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > C
Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
hello sorry, for my late reply my core debugging was at info not at debug, now it's changed and i have the log needed i'm sorry but pastebin doesn't work (it seems that my trace was to big) http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet bigger than 'max_allowed_packet' bytes" i'll send the logfile personal to you, hope you don't dislike this br On 2009-11-24 01:48, Anthony Minessale wrote: > You forgot to set freeswitch to debug loglevel > > You need both of the following: > > console loglevel debug > sofia profile internal siptrace on > > > > > 2009/11/23 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > sorry about wasting your time (wasn't my intent) > > the log is at http://pastebin.freeswitch.org/11240 > i called 5214448370068 (also other calls are in the log) > > they now have changed 180 to 183 on the sonus, but makes no > difference here > > br > > On 2009-11-23 22:07, Anthony Minessale wrote: > > do you have the ringback variable set on the channel? > > if so it will cause 180 to attempt to play inband ringback indication > > > > I have nothing left to say because I asked for the whole log with the > > siptrace enables not just 5 lines of it. > > If you still want help, give me the log to examine and I will > tell you > > what your problem is. > > > > > > > > 2009/11/23 Christian Löschenkohl > <mailto:christian.loeschenk...@xpirio.com> > > <mailto:christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com>>> > > > > thany ou for your answer > > > > we use g729 on all our other connections in passthrough mode > and it > > also doesn't work with alaw. > > so i don't think it's related to this. > > > > br > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > Well its also G729 so I suspect you don't have G729 > > > > > > /b > > > > > > On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: > > > > > >> hi > > >> > > >> our freeswitch server has to talk to a sonus ip-switch > > >> when we want to setup a call we do get a "100 Trying" and then a > > >> "180 Ringing" > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > > >> freeswitch > > >> quits with a CANCEL message. > > >> i simply don't get why our freeswitch aborts the session - i think > > >> it would work > > >> if no "a=sendonly" would be present in the sdp. > > >> > > >> my technical contact doesn't want to switch 180 to 183 on the > sonus > > >> side - this would > > >> also work (i think). in fact he says that 180 ringing is vaild, he > > >> isn't that wrong in > > >> this case. > > >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > >> > > >> 92.63.208.36 - freeswitch > > >> 38.105.229.100 - sonus > > >> > > >> br > > >> > > >> -- > > >> Ing. Christian Löschenkohl > > >> Technische Leitung, Forschung& Entwicklung VoIP > > >> > > >> xpirio > > >> Telekommunikation& Service GmbH > > >> Lakeside B04 > > >> 9020 Klagenfurt > > >> Austria > > >> > > >> T +43 (0) 5 77 11 - 1000 > > >> F +43 (0) 5 77 11 - 1002 > > >> E christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > > <mailto:christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com>> > > >> > > >> ___ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > <mailto:FreeSWITCH-users@lists.
Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
works now thank you very much On 2009-11-27 20:03, Anthony Minessale wrote: > please update to latest trunk 15698 or greater and re-test. > The 183 from the provider had a sendonly attr that tricked the proxy > code into thinking it was a hold/unhold operation. > > > On Fri, Nov 27, 2009 at 10:57 AM, Anthony Minessale > mailto:anthony.miness...@gmail.com>> wrote: > > or you can put it at a url on your web site and just post a link > > > 2009/11/27 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > hello > > sorry, for my late reply > my core debugging was at info not at debug, now it's changed and > i have the log needed > > i'm sorry but pastebin doesn't work (it seems that my trace was > to big) > http://pastebin.freeswitch.org/11305 says "Query failure: Got a > packet bigger than 'max_allowed_packet' bytes" > > i'll send the logfile personal to you, hope you don't dislike this > > br > > On 2009-11-24 01:48, Anthony Minessale wrote: > > You forgot to set freeswitch to debug loglevel > > > > You need both of the following: > > > > console loglevel debug > > sofia profile internal siptrace on > > > > > > > > > > 2009/11/23 Christian Löschenkohl > <mailto:christian.loeschenk...@xpirio.com> > > <mailto:christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com>>> > > > > sorry about wasting your time (wasn't my intent) > > > > the log is at http://pastebin.freeswitch.org/11240 > > i called 5214448370068 (also other calls are in the log) > > > > they now have changed 180 to 183 on the sonus, but makes no > > difference here > > > > br > > > > On 2009-11-23 22:07, Anthony Minessale wrote: > > > do you have the ringback variable set on the channel? > > > if so it will cause 180 to attempt to play inband ringback > indication > > > > > > I have nothing left to say because I asked for the whole > log with the > > > siptrace enables not just 5 lines of it. > > > If you still want help, give me the log to examine and I will > > tell you > > > what your problem is. > > > > > > > > > > > > 2009/11/23 Christian Löschenkohl > > <mailto:christian.loeschenk...@xpirio.com> > > <mailto:christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com>> > > > <mailto:christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com> > > <mailto:christian.loeschenk...@xpirio.com > <mailto:christian.loeschenk...@xpirio.com>>>> > > > > > > thany ou for your answer > > > > > > we use g729 on all our other connections in passthrough > mode > > and it > > > also doesn't work with alaw. > > > so i don't think it's related to this. > > > > > > br > > > > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > > Well its also G729 so I suspect you don't have G729 > > > > > > > > /b > > > > > > > > On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: > > > > > > > >> hi > > > >> > > > >> our freeswitch server has to talk to a sonus ip-switch > > > >> when we want to setup a call we do get a "100 Trying" > and then a > > > >> "180 Ringing" > > > >> within the "180 Ringing" we get a sdp with "a=sendonly" > then our > > > >> freeswitch > > > >> quits with a CANCEL message. > > > >> i simply don't get why our freeswitch aborts the
[Freeswitch-users] mod_php needed
hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian Löschenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_php needed
hi thank you very much for your input i can say for me that i realy tried hard to use the event socket library, but untill now i can't use it like i used all the agi scripts or even mod_perl now. what i do most - in examples, if the server get's an incomming call - find the right user for the number (not that easy because of did in austria), from database or file - build the right dial string for the bridge application (here i miss all the php string functions most) - unsing mod_php functions like setVariable, getVariable, answer, transfer, sleep (i don't see how to do this with the php esl) - or i check if the number is part of a conferencing product and build the right conference setup i think this would also be possible with lua and luasql, but i developed years with phpagi und i'm very used to php in every kind of scripting or how-to-get-a-solution situation (since over 10 years now). for me in our setup it's also the highest goal to get the servers mostly independent of each other. i think nobody of our costumers should be unreachable because a central scripting/event server or also database server has gone away (as developers this happens more often as we would like it to :-)) do not get me wrong, freeswitch is very powerfull and in the near future it will replace nearly all of our asterisk servers. in combination with php the freeswitch plattform would be heaven for me i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php), in combination of the mod_python rewrite (page was last modified in june 2007). br On 2009-06-14 01:15, Nik Middleton wrote: > I couldn’t agree more. We’re working with a group that are developing a > massive PHP based music application. They are experts in PHP and MySQL > but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP > to communicate with the FS event socket, allows them to work on the > areas they know best and not worry about the telephony side too much. > We went the lua route, and don’t use the dial plan at all. My view is > to keep all db access and processing out of FS as much as possible. With > the event socket you simply don’t need to embed anything apart from the > essentials. > > We are now processing 100,000+ call setups a day (4 hours) per server > all using php scripts to drive the application. We may well ultimately > use C++ instead of PHP for the event socket comms, but right now PHP > does just fine. > > Regards > > > > *From:* freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of > *Michael Collins > *Sent:* 13 June 2009 21:57 > *To:* freeswitch-users@lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_php needed > > Perhaps you should look at controlling calls via the FreeSWITCH event > socket instead of from the dialplan. The nice thing about the event > socket is that your call control can happen on a separate machine. There > is a PHP module for the ESL (event socket library) and it would be > relatively easy for you to get going. Here are some links to get you > started: > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > http://wiki.freeswitch.org/wiki/Event_Socket > > If you absolutely MUST have call control with scripts inside of the > dialplan then there simply is no better choice than Lua. You can learn > Lua in a few hours, but getting mod_php finished and debugged will take > time, money, and other resources that no one seems willing to spend. > Here is some information to consider: > > http://wiki.freeswitch.org/wiki/Mod_lua > > Come join us on IRC (#freeswitch on irc.freenode.net > <http://irc.freenode.net>) if you want to discuss this further. > > -MC (IRC: mercutioviz) > > 2009/6/13 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>> > > hello > > i am working for an austrian voip carrier. > for a few months i work with freeswitch and it is simply great. > it solves our needs in many places (high volume, flexible, stable). > the only thing i really miss is the avalibilty of php as a call control > language. > mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't > that good (or even there :-) ). > i know there is perl, i also implemented some applications (conference > system with provisioning, > inbound call routing to our application servers, some tests as pbx), but > what should i say - > perl and me aren't compatible in the end. > > the greatest thing for us would be that mod_php comes alive again with > the functional state > of mod_perl > (http://wiki.freeswitch.org/wiki/Mod_perl_function
Re: [Freeswitch-users] mod_php needed
i tried to
i think i tried everthing and looked closely to everything in libs/esl/php (of
course i build it and included the ESL.php file)
but i do not get the idea in complete, does i work in a client-server way or
in inbound mode like i want to (that is exactly my point)
no examples are there (i would put them in the wiki if i had one)
some simple code i would expect wot work, but i doesn't
execute("setVariable", "codec_string=PCMA");
$esl->execute("answer");
$esl->execute("sleep", "2");
$esl->execute("streamFile",
"/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello");
$esl->execute("hangup", "16");
?>
can you please help me, what do i get wrong?
br
On 2009-06-15 15:59, Anthony Minessale wrote:
> Did you actually use ESL with the php wrapper when you tried?
> You can do all those things from outbound event socket fairly easily.
>
> That mod_php you saw, never worked it was just a stub and it didn't
> actually ever work
> when the guy who added it totally disappeared, I removed it from tree.
>
> And you can still do event socket over localhost on the same box if you
> so choose.
>
> If you really want a mod_php it's entirely possible but it would
> probably cost you upwards
> of 5k in development costs.
>
>
> 2009/6/15 Christian Löschenkohl <mailto:christian.loeschenk...@xpirio.com>>
>
> hi
>
> thank you very much for your input
> i can say for me that i realy tried hard to use the event socket
> library,
> but untill now i can't use it like i used all the agi scripts or
> even mod_perl now.
>
> what i do most - in examples, if the server get's an incomming call
>
> - find the right user for the number (not that easy because of did
> in austria),
>from database or file
> - build the right dial string for the bridge application (here i
> miss all the php
>string functions most)
> - unsing mod_php functions like setVariable, getVariable, answer,
> transfer, sleep
>(i don't see how to do this with the php esl)
> - or i check if the number is part of a conferencing product and
> build the right
>conference setup
>
> i think this would also be possible with lua and luasql, but i
> developed years with
> phpagi und i'm very used to php in every kind of scripting or
> how-to-get-a-solution
> situation (since over 10 years now).
>
> for me in our setup it's also the highest goal to get the servers
> mostly independent
> of each other. i think nobody of our costumers should be unreachable
> because a central
> scripting/event server or also database server has gone away (as
> developers this happens
> more often as we would like it to :-))
>
> do not get me wrong, freeswitch is very powerfull and in the near
> future it will replace
> nearly all of our asterisk servers.
>
> in combination with php the freeswitch plattform would be heaven for me
>
> i also thought Brian Fertig has some source written (as posted on
> http://wiki.freeswitch.org/wiki/Mod_php),
> in combination of the mod_python rewrite (page was last modified in
> june 2007).
>
> br
>
>
> On 2009-06-14 01:15, Nik Middleton wrote:
> > I couldn’t agree more. We’re working with a group that are
> developing a
> > massive PHP based music application. They are experts in PHP and
> MySQL
> > but not in VOIP/Telephony. By tuning an abstraction layer that
> uses PHP
> > to communicate with the FS event socket, allows them to work on the
> > areas they know best and not worry about the telephony side too much.
> > We went the lua route, and don’t use the dial plan at all. My
> view is
> > to keep all db access and processing out of FS as much as
> possible. With
> > the event socket you simply don’t need to embed anything apart
> from the
> > essentials.
> >
> > We are now processing 100,000+ call setups a day (4 hours) per server
> > all using php scripts to drive the application. We may well
> ultimately
> > use C++ instead of PHP for the event socket comms, but right now PHP
> > does just fine.
> >
> > Regards
> >
> >
>
> >
> > *From:* freeswitch-users-boun...@lists.freeswitch.org
> <mailto:freeswitch-users-boun...@lists.freeswitch.org>
>
Re: [Freeswitch-users] mod_php needed
hi
could you provide me a simple example?
- connect with esl
- get uuid
- set a variable (e.g. codec_string=PCMA)
- answer the channel
- playback a file
the script ist called from ivrd, if i get it right
in the dialplan it's
with ivrd started as ./ivrd -h 127.0.0.1 -p
in my setup $esl->api("help") works and also $esl->sendRecv("api help")
but $esl->execute() does nothing
i use version 1.0.4pre8 if it is helpfull
br
On 2009-06-15 17:12, William Suffill wrote:
> Any suggestions of what would be a good example in PHP using ESL to
> document? I'll take a stab at writing something up this week but it
> would help to have some idea what would be useful. I've used it and got
> it working but rather document a generic real life example versus my
> unique use cases.
>
> -- W
>
>
>
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
--
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP
xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
___
Freeswitch-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[Freeswitch-users] channel variable sip_to_tag
hello do someone know how to get the sip_to_tag from an active call? the sip_from_tag is available as a channel variable but sip_to_tag isn't. i don't know if it is available at call setup, the fist time i see the tag=... in the sip header is the challenge response answer from fs i need this to get my aoc (advice-of-charge) implementation running, this one is based on sip info messages and has to contain the same tag's as the active call. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] channel variable sip_to_tag
hi thank you for your reply how can we procced? br On 2009-06-23 18:20, Michael Jerris wrote: > if you need to use the same tags, we should be using the whole same nh > in the code. There is code to do this by call uuid but I can't recall > if thats for NOTIFY or INFO. If its the wrong one, we should add teh > same for what you need. > > Mike > > On Jun 21, 2009, at 6:05 AM, Christian Löschenkohl wrote: > >> hello >> >> do someone know how to get the sip_to_tag from an active call? >> the sip_from_tag is available as a channel variable but sip_to_tag >> isn't. >> i don't know if it is available at call setup, the fist time i see >> the tag=... >> in the sip header is the challenge response answer from fs >> >> i need this to get my aoc (advice-of-charge) implementation running, >> this one >> is based on sip info messages and has to contain the same tag's as >> the active call. >> >> br >> >> -- >> Ing. Christian Löschenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenk...@xpirio.com >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation
hi i'm inalp patton certified, so maybe i can help if you can post your config (export startup config). please also describe your setup if i could get http+telnet access to the smartnode, the system would be registered in an minute br On 2009-06-29 17:11, Frederik Denkens wrote: > Hi, > > Following the recommendations of this list, we went with an external > gateway to connect to the BRI based ISDN network. > > We'd like to configure a Patoon 4554 2xBRI<-> SIP gateway with > Freeswitch to connect to the public ISDN network. The freeswitch > config seems quite straightforward, but we don't manage to get the > Patton to register with the Freeswitch and vise versa. > > Does anybody have some insight/sample config/tips they can share with > us on this? Honesty obliges me to say that our experience with the > Patton product is quite limited, which is not a big help for such a > complex product. > > Many thanks in advance! > > Frederik. > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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
