[Freeswitch-users] Monitoring IVR pressed-options in XML IVR
Hi,
I am currently creating IVR using the functions provided in the XML dialplan
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
Using functions like this
entry action=menu-play-sound digits=1
param=$${base_dir}/1255549537_Welcome.wav/
I can play files, etc.
I wonder what is the smartest way to monitor (as in big brother) the
options selected by the user:
I assume that I can include an entry of the type:
entry action=menu-exec-app digits=1 param=javascript foo.js/
and include in foo.js the code to track the selection.
But I wonder if this is the best approach
/aep
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Re: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR
I want to trigger CUSTOM events via ESL as they navigate inside of the IVR.
The XML IVRs are generated from a GUI.
The CUSTOM events need to carry
- what IVR the user is navigating
- what option has been selected
- ideally how long they stayed listening (this can be calculated)
- and when they hang the phone
/aep
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Stopping junk mailers is good for the environment
On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero aep.li...@it46.se
wrote:
Hi,
I am currently creating IVR using the functions provided in the XML
dialplan
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
Using functions like this
entry action=menu-play-sound digits=1
param=$${base_dir}/1255549537_Welcome.wav/
I can play files, etc.
I wonder what is the smartest way to monitor (as in big brother) the
options selected by the user:
I assume that I can include an entry of the type:
entry action=menu-exec-app digits=1 param=javascript foo.js/
and include in foo.js the code to track the selection.
But I wonder if this is the best approach
/aep
Are you trying to do some sort of live monitoring as it happens (i.e.
while
the call is live) or do you just want a record of the digits they pressed?
-MC
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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?
You can use the api and check that the channel is occupied with show channels? You can write a small javascript that checks if the channel is occupied by means of session.execute api. /aep -- Stopping junk mailers is good for the environment My SIP provider allows only one call (incoming or outgoing) via one SIP account. For FreeSWITCH I have configured it as public DID extension and outgoing gateway. Now I would like to transfer to another gw (or generate limit exceded) when one tries to place an outgoing call while incoming call is in progress. How tho do that? Limiting the number of outgoing calls is easy (mod_limit), but how to take into account incoming one? - Dmitry Bely ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Dynamic volume adjustment in heterogenous IVR menus
Sorry for the email subject that sounds like a IEEE paper. I am building IVRs using FS API and sending out audio that is a combination of TTS and playing WAV files. What is the best way to control volume levels? I know i might be asking for magic here... In any case, is there any simple ways to add gain to certain nodes of am IVR? /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Bind to more than one ethernet interface
If I am correct you need to create a sip profile per interface and hardcode/set the IP address of each interface correctly in the SIP RTP fields of the profile. Then you need to set carefully the correct NAT and auth options for each profile /aep -- Stopping junk mailers is good for the environment Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either auto or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] checking subscribed/subscribers to events
Hi, Is there any simple way to know: who is subscribed to certain events via ESL? check which events i have subscribed during a ESL session? control which events can one user subscribe? disable the subscription of certain events and not all at the same time? /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Status of ubuntu/debian packages.
Hi, Now I seem to reach the webserver. How do i checkout a local copy to run the builder? /aep -- Stopping junk mailers is good for the environment It seems I had a port forwarded incorrectly for the external access to the git web interface. here it is again: http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/ I've tested it to work now. -William King Hadley Rich wrote: On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote: Currently it's /opt/freeswitch. I would like to see it move to FHS correct locations for inclusion in to debian/ubuntu. This is the next bit that I will be working on. Yeah, the FHS stuff was the bit that I got a little stuck on a while back. Of course we also hope that the debian voip team will pick it up once we've cleaned it up. Sounds good. I am not an ubuntu guy so I can't speak to that. I would say that most of the licenses of the included packages would allow for inclusion in debian main. Things like the cepstral support would have to go in to contrib. Gotcha, multiverse is for not free software, so anything that can go into main in Debian could go into universe in Ubuntu. hads ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Call Files for a dialer engine
I am exploring the possibility of building a Dialer that emulates the logic of Call Files in asterisk. A CallerID catcher is creating CUSTOM events that I can store in a database. I can trigger callbacks using ESL but I wonder what is the best way/nicer/geekier to do something like outgoing calls in * /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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-dev] Status of ubuntu/debian packages.
Hi William, I will be very happy to test them, can you share the source and procedure to create the .debs? It will be also very good to find ways to have a cepstral package included *pending the licence* of course :) /aep -- Stopping junk mailers is good for the environment Just to give everyone an update. There are working Ubuntu packages in a launchpad ppa. Debian users can add the ppa to their apt sources and build the package on your box. I'm currently using the packages on my home box and it is working great. Alright. I'm looking for people who want to use the packages. There are built packages for Ubuntu 8.04, 8.10, 9.04, and I'm working on 9.10 as well. I'm building two apt repos. One will have nightly builds and the other will be for tagged releases, plus any major bug fixes. Thanks to Frank, we now have everything split out into separate files for everything. This is to try to reduce the amount of stuff you 'have' to download by default. We have the en-us-callie sounds packaged at 8k, 16k, 32k, and 48k, we also have packaged the russian-elena and music on hold at the same qualities. If there are other languages, or voices I'd be more than happy to package them. I just need the 48k. Also we have separated the packages out so you can specify which mods you want, such as mod_perl, mod_python, mod_lua are all separate so you can install them if you want. nightlies: https://launchpad.net/~pbxbuntu-drivers/+archive/ppa tagged releases: https://launchpad.net/~freeswitch-drivers/+archive/ppa The tagged released packages will start with the 1.0.5 tagged release which some say should be coming out soon. Any one who would like to help out feel free to sign up with a launchpad account, and request to join the driver team of either or both ppa's. Once we have some people using the packages, and testing them, I have already talked to some of the ubuntu official package maintainers about what would need to be done to add freeswitch into the ubuntu multiverse repo. Any questions? -William King (quentusrex) ___ FreeSWITCH-dev mailing list freeswitch-...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev 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 Files for a dialer engine
Yes, sounds the best way to go. I assume that Unique-ID is the unique key to track the call via ESL Unique-ID: a984afd4-a865-11de-a5b4-fb5a867b002c and Answer-State: the variable to determine if the call is successful? Or should wait for the reason of CS_DESTROY message. I want to avoid to keep track of the whole state machine to know if a call has been completed successfully or not. /aep Unique-ID: 53f51090-a865-11de-a5b4-fb5a867b002c Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered -- Stopping junk mailers is good for the environment make an esl script that monitors a dir for new files, and push the contents into your same db? On Wed, Sep 23, 2009 at 10:32 AM, Alberto Escudero aep.li...@it46.sewrote: I am exploring the possibility of building a Dialer that emulates the logic of Call Files in asterisk. A CallerID catcher is creating CUSTOM events that I can store in a database. I can trigger callbacks using ESL but I wonder what is the best way/nicer/geekier to do something like outgoing calls in * /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 mailing list FreeSWITCH-users@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] Callback in Javascript, session.destroy() does not free the channel!
Hi Michael,
I will like to get a few RINGS back to the user and sleep a bit before the
call back.
The second i can do using the app sleep.
What about the first thing?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
Will test i let you know...
Crazy Callbacker aka aep
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FYI,
I did a POC on this:
extension name=crazy_callback
condition field=destination_number expression=^(\d{10})$
action application=set data=luarun dump_arg.lua
${caller_id_number}
${caller_id_name}/
/condition
/extension
dump_arg.lua:
--
dump_args.lua
-- print out the
args
freeswitch.consoleLog(info, Arg1: .. argv[1] ..
\n)
freeswitch.consoleLog(info, Arg2: .. argv[2] ..
\n)
From there you can do whatever you want in the target script. I'm sure
perlrun, pyrun, and jsrun are all the same in terms of accepting args and
running whatever you want, like generating an originate API, etc. Just
remember that the caller needs to hangup before you can call him back. :)
-MC
On Fri, Sep 18, 2009 at 7:53 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
You could put an api_hangup_hook on the channel to jsrun your script.
What you want with javascript is not going to happen as long as you
execute
the script *WITH* the channel.
it's not a problem it's just misuse/misunderstanding on your part.
On Fri, Sep 18, 2009 at 5:03 AM, Alberto Escudero
aep.li...@it46.sewrote:
Thanks for all the tips. I tried to run apiExecute(bgapi, jsrun
originate)
and still Javascript?? does not let the thread go.
No matter the combination of session.hangup(), exit, apiExecute with or
without bgapi, the state remains in CS_EXECUTE.
So at the end i am triggering an event that i can later use to execute
a
originate callback. It is nicer with ESL but i still think that will be
nice to have a real way to expunge a second Javascript and let the
first
one die.
The GSM channel/modem needs to be free-free (as I am a serial
port-free)
to handle the outgoing call. The callback script worked perfect with
SIP
because it does not care how many sessions are running in parallel. It
can
always place a call back event the channel is not properly close.
/aep
--
Stopping junk mailers is good for the environment
So, what happens is that when you are executing an app, the state is
CS_EXECUTE. Even if the session is hungup, the state machine doesn't
go
through all the hangup code until your app executes.
The easiest workaround is probably to start a background api (bgapi?)
call
to a script. This will happen on another thread, then allow your
current
thread to execute and the hangup code will execute. This should work
just
fine, I think. (You can stop reading here.)
But wait, there's even more fun! anthm recently checked in a change a
couple days that lets you work around this. Don't call destroy, call
hangup on the session, on that session's thread. This will perform a
hangup, then progress the state machine. Then the session will truly
be
hungup. Maybe you need update your freeswitch code, if this is not
happening for you.
If you updated and hangup still isn't hanging up, you might want to
ask
specifically about that. Or, you may need to call
switch_core_session_hangup_state directly -- just hangup alone might
not
do the trick. This is a C function, and not exposed to languages by
default - you can either patch javascript plugin to expose this
safely
(and I have no idea what this means for the javascript runtime), or
use
a
more capable plugin like mod_managed which _does_ expose all the C
functions, and lets you call in and out of them as you please.
And now, someone who knows what they're talking about will chime in
and
point out what I got wrong.
Thanks,
-Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Alberto Escudero
Sent: Thursday, September 17, 2009 3:20 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Callback in Javascript, session.destroy()
does
not free the channel!
We are trying to create a callback application in Javascript. We get
the
callerid from the unanswered call and after destroying the session,
we
initiate a callback to the user to conenct it to a local extension in
the
dialplan.
Although we have tried to destroy the first session, or even invoke a
second script using apiExecute(jsrun,dialer.js), tried
session.hangup()
or exit()... the first session does not seem to close properly until
the
whole chain of scripts are completed.
Here is a piece of code that shows the concept (yes!, the sleep
function
is far from ideal. CPU loves it! )
function sleep(milliseconds) {
var start = new Date().getTime();
for (var i = 0; i 1e7; i++) {
if ((new Date
Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!
Thanks for all the tips. I tried to run apiExecute(bgapi, jsrun originate)
and still Javascript?? does not let the thread go.
No matter the combination of session.hangup(), exit, apiExecute with or
without bgapi, the state remains in CS_EXECUTE.
So at the end i am triggering an event that i can later use to execute a
originate callback. It is nicer with ESL but i still think that will be
nice to have a real way to expunge a second Javascript and let the first
one die.
The GSM channel/modem needs to be free-free (as I am a serial port-free)
to handle the outgoing call. The callback script worked perfect with SIP
because it does not care how many sessions are running in parallel. It can
always place a call back event the channel is not properly close.
/aep
--
Stopping junk mailers is good for the environment
So, what happens is that when you are executing an app, the state is
CS_EXECUTE. Even if the session is hungup, the state machine doesn't go
through all the hangup code until your app executes.
The easiest workaround is probably to start a background api (bgapi?) call
to a script. This will happen on another thread, then allow your current
thread to execute and the hangup code will execute. This should work just
fine, I think. (You can stop reading here.)
But wait, there's even more fun! anthm recently checked in a change a
couple days that lets you work around this. Don't call destroy, call
hangup on the session, on that session's thread. This will perform a
hangup, then progress the state machine. Then the session will truly be
hungup. Maybe you need update your freeswitch code, if this is not
happening for you.
If you updated and hangup still isn't hanging up, you might want to ask
specifically about that. Or, you may need to call
switch_core_session_hangup_state directly -- just hangup alone might not
do the trick. This is a C function, and not exposed to languages by
default - you can either patch javascript plugin to expose this safely
(and I have no idea what this means for the javascript runtime), or use a
more capable plugin like mod_managed which _does_ expose all the C
functions, and lets you call in and out of them as you please.
And now, someone who knows what they're talking about will chime in and
point out what I got wrong.
Thanks,
-Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Alberto Escudero
Sent: Thursday, September 17, 2009 3:20 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does
not free the channel!
We are trying to create a callback application in Javascript. We get the
callerid from the unanswered call and after destroying the session, we
initiate a callback to the user to conenct it to a local extension in the
dialplan.
Although we have tried to destroy the first session, or even invoke a
second script using apiExecute(jsrun,dialer.js), tried session.hangup()
or exit()... the first session does not seem to close properly until the
whole chain of scripts are completed.
Here is a piece of code that shows the concept (yes!, the sleep function
is far from ideal. CPU loves it! )
function sleep(milliseconds) {
var start = new Date().getTime();
for (var i = 0; i 1e7; i++) {
if ((new Date().getTime() - start) milliseconds){
break;
}
}
}
if (session.ready()) {
//We catch the caller_id
caller_id_num = session.caller_id_num;
console_log(Now we got your Caller ID\n);
//How long we want to wait to trigger a call back
session.execute(sleep,5000);
console_log(We have waited a while... time to create the
callback\n);
//apiExecute(jsrun, callback.js);
}
//Destroy the session...
session.destroy();
session=undefined;
sleep(1);
//Preparing callback
session2 = new
Session('{ignore_early_media=true}celliax/interface1/600464646');
session2.setAutoHangup(false);
session2.answer();
exit();
++
Wisdom thoughts?
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Re: [Freeswitch-users] Not able to make call using external profile
Have you tried with
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
instead?
/aep
--
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hi folks, I m not able to make SIP calls using external profile.
i have added the following lines to the
$installdir/conf/dialplan/public.xml
extension name=echo
condition field=destination_number expression=^9996$
action application=answer/
action application=echo/
/condition
/extension
extension name=public_extensions
condition field=destination_number expression=^(10[01][0-9])$
action application=bridge data=sofia/external/$...@$${domain}/
/condition
/extension
I m able to connect using 1000 and 1001 from public Internet. I am able
to
make an echo call.
*when i type :*
$: sofia status profile external reg
It shows the list of the connected clients and their information.
but when I m trying to make a call from one user to other user, it
generates
the following error
2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel
sofia/external/1...@192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be]
2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing
1000-1000 in context public
2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel
sofia/external/1...@192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306]
2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway
found
2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup
sofia/external/
1...@192.168.1.50 [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING]
2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: MANDATORY_IE_MISSING
2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup
sofia/external/1...@192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING]
2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1
(sofia/external/1...@192.168.1.50) Ended
2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close
Channel
sofia/external/1...@192.168.1.50 [CS_DESTROY]
2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2
(sofia/external/1...@192.168.1.50) Ended
2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close
Channel
sofia/external/1...@192.168.1.50 [CS_DESTROY]
with regards
Pankaj anand
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[Freeswitch-users] FreeSWITCH and OpenBTS integration
I am happy to let you know that FreeSWITCH route calls from OpenBTS, the open base station based on the Universal Software Radio USRP. Yes! Calls from a standard handset to a GSM base station connected to FreeSWITCH If you want to read more about the idea check: http://openbts.sourceforge.net/ http://www.it46.se/entry/380 (our effort to deploy the technology in a developing region) I have put a few notes for others to give it a try available here: https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH Let me know what is the best place in FreeSWITCH wiki to add and keep updated this information -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 and OpenBTS integration
Sorry, just realized that the sourceforge page is protected by password. I am happy to put the info in FreeSWITCH wiki, where does it make sense to add this project info? -aep -- Stopping junk mailers is good for the environment I am happy to let you know that FreeSWITCH route calls from OpenBTS, the open base station based on the Universal Software Radio USRP. Yes! Calls from a standard handset to a GSM base station connected to FreeSWITCH If you want to read more about the idea check: http://openbts.sourceforge.net/ http://www.it46.se/entry/380 (our effort to deploy the technology in a developing region) I have put a few notes for others to give it a try available here: https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH Let me know what is the best place in FreeSWITCH wiki to add and keep updated this information -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH and OpenBTS integration
The real challenge at the moment is to find adequate regulatory scenarios to run this technology. In many parts of the world where we works operators/governments have literally locked the spectrum so others can not run anything on it. Community networks are left with garbage bands to operate 802.11 devices. We welcome any scenarios where to operate an open gsm infrastructure connected to IP. /aep -- Stopping junk mailers is good for the environment Am Thursday 17 September 2009 schrieb Steve Underwood: On 09/17/2009 09:19 PM, Brian West wrote: Just create an OpenBTS page on our wiki. /b On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: Sorry, just realized that the sourceforge page is protected by password. I am happy to put the info in FreeSWITCH wiki, where does it make sense to add this project info? Isn't there still some legal wrangling over openBTS? Steve nope, that was resolved: http://openbts.blogspot.com/2009/07/three-quotes.html -- --- Stefan Knoblich| Web: http://www.axsentis.de/ axsentis GmbH |http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knobl...@axsentis.de UST-ID: DE244977565| JID: s.knobl...@jabber.axsentis.de --- Web: http://stkn.techmage.de/ Email: s...@freeswitch.org IRC: #freeswitch-de @ irc.freenode.net ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH and OpenBTS integration
Done! http://wiki.freeswitch.org/wiki/OpenBTS -- Stopping junk mailers is good for the environment Just create an OpenBTS page on our wiki. /b On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: Sorry, just realized that the sourceforge page is protected by password. I am happy to put the info in FreeSWITCH wiki, where does it make sense to add this project info? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!
We are trying to create a callback application in Javascript. We get the
callerid from the unanswered call and after destroying the session, we
initiate a callback to the user to conenct it to a local extension in the
dialplan.
Although we have tried to destroy the first session, or even invoke a
second script using apiExecute(jsrun,dialer.js), tried session.hangup()
or exit()... the first session does not seem to close properly until the
whole chain of scripts are completed.
Here is a piece of code that shows the concept (yes!, the sleep function
is far from ideal. CPU loves it! )
function sleep(milliseconds) {
var start = new Date().getTime();
for (var i = 0; i 1e7; i++) {
if ((new Date().getTime() - start) milliseconds){
break;
}
}
}
if (session.ready()) {
//We catch the caller_id
caller_id_num = session.caller_id_num;
console_log(Now we got your Caller ID\n);
//How long we want to wait to trigger a call back
session.execute(sleep,5000);
console_log(We have waited a while... time to create the
callback\n);
//apiExecute(jsrun, callback.js);
}
//Destroy the session...
session.destroy();
session=undefined;
sleep(1);
//Preparing callback
session2 = new
Session('{ignore_early_media=true}celliax/interface1/600464646');
session2.setAutoHangup(false);
session2.answer();
exit();
++
Wisdom thoughts?
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[Freeswitch-users] originate command sofia behaviour
I will like to update the wiki to spell out clearly the differences between this three commands I have a IVR running in 4600 and the FS box has IP address 192.168.46.15 originate sofia/192.168.46.15/1001 4600 originate sofia/internal/1...@192.168.46.15 4600 originate sofia/internal/1001%192.168.46.15 4600 The first originate places a call as a external gateway, not until registered phone 1001 answers the call is transfer to 4600 The second and third originate command triggers extension 4600 Javascript IVR although 1001 has not answer Can anyone clarify me if this is the intended behavior also including the difference between % and @ /aep ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] originate command sofia behaviour
The problem i am facing is the following: Extension 4600 is a Javascript IVR that starts by session.aswer() I want to originate a call to leg 1 and then connected to the IVR when the leg 1 has answered. If I run originate sofia/192.168.46.15/1001 4600 call is transfer to extension 4600 *IVR* after 1001 answers the call If I run originate sofia/internal/1...@192.168.46.15 4600 the IVR starts BEFORE user 1001 has answered? What is the best way to: Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR) after leg 1 has answered the call? /aep -- Stopping junk mailers is good for the environment On Wed, Sep 16, 2009 at 7:26 AM, Alberto Escudero aep.li...@it46.se wrote: I will like to update the wiki to spell out clearly the differences between this three commands I have a IVR running in 4600 and the FS box has IP address 192.168.46.15 originate sofia/192.168.46.15/1001 4600 originate sofia/internal/1...@192.168.46.15 4600 originate sofia/internal/1001%192.168.46.15 4600 The first originate places a call as a external gateway, not until registered phone 1001 answers the call is transfer to 4600 The second and third originate command triggers extension 4600 Javascript IVR although 1001 has not answer Can anyone clarify me if this is the intended behavior also including the difference between % and @ The difference between % and @ is discussed here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] originate command sofia behaviour
Yes, it did work! No we do not need to pay for several GSM calls to test
a IVR script!
/aep and gmaruzz
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On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero
aep.li...@it46.sewrote:
The problem i am facing is the following:
Extension 4600 is a Javascript IVR that starts by session.aswer()
I want to originate a call to leg 1 and then connected to the IVR when
the
leg 1 has answered.
If I run
originate sofia/192.168.46.15/1001 4600
call is transfer to extension 4600 *IVR* after 1001 answers the call
If I run
originate sofia/internal/1...@192.168.46.15 4600
the IVR starts BEFORE user 1001 has answered?
What is the best way to:
Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR)
after leg 1 has answered the call?
You can try ignoring early media to force the A-leg to answer before
anything else happens. Try this and let us know if it does what you want:
originate {ignore_early_media=true} sofia/internal/1...@192.168.46.15 4600
You can probably look at the SIP traces of the two options you've tried
(without ignoring early media) to confirm that you're getting media prior
to
answer when doing originate sofia/internal/1...@192.168.46.15 4600 -
probably in one case you get a 180 and in the other a 183. Check it out
and
let us know. :)
-MC
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Re: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
After digging into this issue, it might the case that the implementation of out-bound DTMF of the client i am using does not properly increments CSeq per DTMF. For those interested, i am currently integrating OpenBTS with Freeswitch! :) -aep -- Stopping junk mailers is good for the environment Hi, I am using the function session.collectInput and session.streamFile to collect a number of DTMF digits. If the DTMF digits are sent in the RTP, i can collect several digits until timeout. No problem there! If the DTMFs are received as a sequence of SIP INFO packages, collectInput only receives the first one. Any ideas? -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
Hi, I am using the function session.collectInput and session.streamFile to collect a number of DTMF digits. If the DTMF digits are sent in the RTP, i can collect several digits until timeout. No problem there! If the DTMFs are received as a sequence of SIP INFO packages, collectInput only receives the first one. Any ideas? -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] wav2mp3 conversion inside of spidermonkey
Hi Steven, Sounds like a very good tip. Do you have any example available to share? I will be happy to upload it to the wiki when i put it up and running. /aep -- Stopping junk mailers is good for the environment I run into this problem before. Don't remember the exact error but might be segfault of lame runing in freeswitch-lua. If you use Linux you would like to try iwatch. It's a perl program watching your file system and can execute the lame command as soon as it got the CLOSE_WRITE(or other) filesystem event. On Aug 26, 2009, at 4:44 PM, Michael Jerris wrote: Running out of stack space? The stack space we run freeswitch in is fairly small. Programs launched from the freeswitch process inherit this. Mike On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote: I ran strace from freeswitch and from the command line. lame segfaults when run from system FS. The only obvious different i see is in the execve() /* XX vars */ apart from the final Segfault From execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/foo.mp3, - S], [/* 16 vars */]) = 0 From FS execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/fooo.mp3, -S], [/* 14 vars */]) = 0 I am attaching the full straces in case they are of any help. Not sure if this deserves a jira /aep -- Stopping junk mailers is good for the environment maybe it's writing some err to stderr that is being suppressed somehow On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) aep.li...@it46.se wrote: Hi Brian, From the CLI freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.wav and running the command from the command line: open46:/tmp# /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -Sopen46:/tmp# ls foo.mp3 foo.wav If I do the same with lame397 freeswi...@open46 system /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.mp3 foo.wav Highly paranormal! Sorry for hijacking the previous thread. /aep -- Stopping junk mailers is good for the environment Try running it at the CLI and see if you see any errors. Also please do not hijack threads. The original thread [Freeswitch-users] XML- RPC on different ip than 0.0.0.0 which was hijacked by clicking reply, changing the subject and clicking send. Please in the future do not do that as it clutters up the threading and could get your query lost in the noise. Thanks, Brian On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote: Here it comes the mystery. I am use lame 3.98.2 the mp3 file never appears, if I use version 3.97 (older version), it does!. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 lame_strace.txt___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] freeswitch compile error
You can always download it and test it for a while!. It will insert some nice commercials when running TTS. I ended up buying a license, Allison convinced me :) And if you plan to use TTS via PSTN or GSM, purchase a 8 Khz voice. /aep -- Stopping junk mailers is good for the environment Unless you have installed the Cepstral SDK you can't compile mod_cepstral. Please visit www.cepstral.com to purchase voices and the SDK is included. /b On Aug 29, 2009, at 5:01 PM, Erwin Huang wrote: mkdir .libs Compiling mod_cepstral.c ... mod_cepstral.c:41:19: error: swift.h: No such file or directory ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Centos 5.3 vs Ubuntu LTS
Debian :) -- Stopping junk mailers is good for the environment On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa te...@voicetechnology.com.br wrote: Hi folks,What linux distro would you suggest to deploy FS with django: Centos 5.3 or Ubuntu LTS? CentOS 5.3 FTW! -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Internal Profile sends RTP to external IP when using two network interfaces
Hi, I have a FS box with two physical network interfaces. The internal interface is hosting several internal phones. I have binded the internal profile SIP/RTP/IP to the private interface. Phones registered correctly but with User: 1...@external.ip.address in sofia status profile internal When I place a call between two internal phones, RTP traffic is send to the external IP address of the FS box instead of the internal. The SIP/SDP messages send from FS carry the external IP instead of the internal. The result is that no RTP media arrives to any of the phones. /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] wav2mp3 conversion inside of spidermonkey
I ran strace from freeswitch and from the command line. lame segfaults
when run from system FS.
The only obvious different i see is in the execve() /* XX vars */ apart
from the final Segfault
From
execve(/usr/local/freeswitch/bin/lame,
[/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/foo.mp3, -S],
[/* 16 vars */]) = 0
From FS
execve(/usr/local/freeswitch/bin/lame,
[/usr/local/freeswitch/bin/lame, /tmp/foo.wav,
/tmp/fooo.mp3, -S], [/* 14 vars */]) = 0
I am attaching the full straces in case they are of any help. Not sure if
this deserves a jira
/aep
--
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maybe it's writing some err to stderr that is being suppressed somehow
On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists)
aep.li...@it46.se wrote:
Hi Brian,
From the CLI
freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2
/tmp/foo.wav
/tmp/foo.mp3 -S
2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing
command:
/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S)] output:
+OK
open46:/tmp# ls
foo.wav
and running the command from the command line:
open46:/tmp# /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3
-Sopen46:/tmp# ls
foo.mp3 foo.wav
If I do the same with lame397
freeswi...@open46 system /usr/local/freeswitch/bin/lame397 -V2
/tmp/foo.wav /tmp/foo.mp3 -S
2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing
command:
/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
/tmp/foo.mp3 -S)] output:
+OK
open46:/tmp# ls
foo.mp3 foo.wav
Highly paranormal! Sorry for hijacking the previous thread.
/aep
--
Stopping junk mailers is good for the environment
Try running it at the CLI and see if you see any errors. Also please
do not hijack threads. The original thread [Freeswitch-users] XML-
RPC on different ip than 0.0.0.0 which was hijacked by clicking
reply, changing the subject and clicking send. Please in the future
do not do that as it clutters up the threading and could get your
query lost in the noise.
Thanks,
Brian
On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
appears, if I use version 3.97 (older version), it does!.
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FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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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
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FROM COMMAND LINE
==
[1]+ Stopped /usr/local/freeswitch/bin/freeswitch (wd:
/usr/local/freeswitch/conf)
(wd now: /usr/local/freeswitch/bin)
.wav /tmp/foo.mp3 -Seeswitch/bin# strace /usr/local/freeswitch/bin/lame
/tmp/foo.
execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame,
/tmp/foo.wav, /tmp/foo.mp3, -S], [/* 16 vars */]) = 0
brk(0) = 0xa012000
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or directory)
mmap2(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xb7f5f000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=63289, ...}) = 0
mmap2(NULL, 63289, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f4f000
close(3)= 0
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or directory)
open(/lib/libncurses.so.5, O_RDONLY) = 3
read(3,
\177ELF\1\1\1
Re: [Freeswitch-users] Internal Profile sends RTP to external IP when using two network interfaces
ASCII art follows voip-phones. 10.0.46.1 [ FS ] 216.82.231.69 Internet I have two profiles binded to two different IPs internal is binded to 10.0.46.1 port 5060 /24 network external is binded to 216.82.231.69 port 5080 My IP phones are in the 10.0.46.X range and they register @ 10.0.46.1 IP address. When I place a call between the two phones in the 10.0.46.0/24 network the phones register but audio RTP is sent to 216.82.231.69 instead of 10.0.46.1 (FS) In this scenario i have FS without any NAT configuration. -- The way that i solved was to run both profiles (internal and external) in the public IP (local_ipv4) and force the phones to register with 216.82.231.69 instead of 10.0.46.1 Thanks looong time! -- Stopping junk mailers is good for the environment First off you can't bind one profile to two interfaces you have to launch two sofia profiles, one for each IP. Secondly if you're doing things like this you'll have to refer to the in tree internal.xml. Third can you outline the network topology a little bit more? Is nat involved? /b On Aug 26, 2009, at 3:04 AM, Alberto Escudero-Pascual (lists) wrote: Hi, I have a FS box with two physical network interfaces. The internal interface is hosting several internal phones. I have binded the internal profile SIP/RTP/IP to the private interface. Phones registered correctly but with User: 1...@external.ip.address in sofia status profile internal When I place a call between two internal phones, RTP traffic is send to the external IP address of the FS box instead of the internal. The SIP/SDP messages send from FS carry the external IP instead of the internal. The result is that no RTP media arrives to any of the phones. /aep ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Internal Profile sends RTP to external IP when using two network interfaces
Yes, I did fix the internal IP in the internal profile overwritting the local_ipv4 var in sip-ip and rtp-ip. When the internal clients sofia status profile internet User: 1...@216.82.231.69 Contact:user sip:1...@10.0.46.51;user=phone;fs_nat=yes;fs_path=sip%3A1000%4010.0.46.51%3A5060%3Buser%3Dphone Agent: Grandstream BT110 1.0.8.12 Status: Registered(UDP-NAT)(unknown) EXP(2009-08-26 22:22:32) Host: open46 IP: 10.0.46.51 Instead of User: 1...@10.0.46.51 I can see SIP/SDP messages announcing the external IP and port 5060 in the INVITE. It is a very old phone... will try with different gadgets tomorrow. /aep -- Stopping junk mailers is good for the environment Well actually you do have issues you're going to have to specify the rtp-ip and sip-ip on BOTH profiles yourself because you're config is currently putting your external IP into the internal's settings. Please correct that and I'm sure it'll work then. /b On Aug 26, 2009, at 9:18 AM, Alberto Escudero-Pascual (lists) wrote: ASCII art follows voip-phones. 10.0.46.1 [ FS ] 216.82.231.69 Internet I have two profiles binded to two different IPs internal is binded to 10.0.46.1 port 5060 /24 network external is binded to 216.82.231.69 port 5080 My IP phones are in the 10.0.46.X range and they register @ 10.0.46.1 IP address. When I place a call between the two phones in the 10.0.46.0/24 network the phones register but audio RTP is sent to 216.82.231.69 instead of 10.0.46.1 (FS) In this scenario i have FS without any NAT configuration. -- The way that i solved was to run both profiles (internal and external) in the public IP (local_ipv4) and force the phones to register with 216.82.231.69 instead of 10.0.46.1 Thanks looong 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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] wav2mp3 conversion inside of spidermonkey
Dear all, In one of the applications I am writing I need to convert a recorded wav to mp3. After using session.recordFile() and obtaining a foo.wav file, I am calling session.execute(system,lmLameCmd); to invoke lame for the conversion. The system command looks like this: lmLameCmd = /usr/local/freeswitch/bin/lame -V2 foo.wav foo.mp3 -S; Here it comes the mystery. I am use lame 3.98.2 the mp3 file never appears, if I use version 3.97 (older version), it does!. If I execute the conversion from the command line, i get the mp3 with both 3.97 and 3.98.2 In fact, i am considering doing the conversions as background job, but I am very curious to hear if this behavior has a pseudo-scientific explanation /aep -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] wav2mp3 conversion inside of spidermonkey
Hi Brian, From the CLI freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.wav and running the command from the command line: open46:/tmp# /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -Sopen46:/tmp# ls foo.mp3 foo.wav If I do the same with lame397 freeswi...@open46 system /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.mp3 foo.wav Highly paranormal! Sorry for hijacking the previous thread. /aep -- Stopping junk mailers is good for the environment Try running it at the CLI and see if you see any errors. Also please do not hijack threads. The original thread [Freeswitch-users] XML- RPC on different ip than 0.0.0.0 which was hijacked by clicking reply, changing the subject and clicking send. Please in the future do not do that as it clutters up the threading and could get your query lost in the noise. Thanks, Brian On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote: Here it comes the mystery. I am use lame 3.98.2 the mp3 file never appears, if I use version 3.97 (older version), it does!. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
