-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Michael,
on my ubuntu 8.04 I have libtiff4 and libtiff4-dev installed. libtiff
and libtiff-dev is not installed. I gonna test it today
regards
helmut
Michael Collins schrieb:
> Helmut,
> I think Mike J was pointing out that spandsp needs libtiff
Hi Erick,
Not sure if you've tried this (or if it'll help), but you can force
routing in the dialplan like so:
Cheers --
Dave
i forgot to give you the pastebin URL
http://pastebin.freeswitch.org/6379
I'm running latest trunk - Revision: 10682
I've been doing an ngrep on my external
Cable modem <> nat router <> fs
fs is set as DMZ on nat router so all packets get there.
My ipv4 address is 192.168.0.x The nat router holds the public IP. Public
IP is a registered domain sparkz.tv so addressable from the internet cloud.
Since fs is DMZ, all requests for sparkz.tv o
I can offer a bit of a bounty for this. Can anyone else chip in?
Thanks -
JP
On Tue, Dec 9, 2008 at 11:45 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:
> It is something we have been discussing as we need these stats to do
> rtcp properly but we have not written any code to do so. It is
> "som
Thanks Anthony , you did a great work ! this is fixed in svn r10691.
Some notes for people using Sonus and L3 as was my case :
in var.xml in some scenario you may need :
in sip_profiles/internal.xml :
might help for some people with rtp issues :
If you have issues with DTMF and timestamp
Pretty simple...
-Original Message-
Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC
What do your records say? Ie do they balance to what the carrier
claims? You should at a minimum have macro level data to confirm
against.
27% seems high, but even at that level if you assume your remaining
population is "normal" you are still no where close to call center /
predictive tr
Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC
On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <[EMAIL PROTECTED]> wrote:
> On our last bill, the carrier said we had 27% short duration calls (maybe
> t
On our last bill, the carrier said we had 27% short duration calls
(maybe they are wrong but it was on the bill). It is definitely not
call center. But these callers hangup as soon as they hear answer
machine or most of the time a ring back tone from cell phone. This
class of caller will call a c
Curious if anyone has practical real world input on training CMU based
ASR engines (Sphinx, PocketSphinx) and / or creating and tuning voices
for the TTS related components.
Just trying to understand how hard it is, what the realistic gap is to
use these tools in real world applications.
i forgot to give you the pastebin URL
http://pastebin.freeswitch.org/6379
>
> I'm running latest trunk - Revision: 10682
>
> I've been doing an ngrep on my external freeswitch SIP port and FS
> is not sending any SIP packets anywhere when I run the following command.
> Bumping up TPORT_LOG to 9 a
I'm running latest trunk - Revision: 10682
I've been doing an ngrep on my external freeswitch SIP port and FS
is not sending any SIP packets anywhere when I run the following command.
Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are logged.
originate
'sofia/external/[EMAIL PRO
originate 'sofia/internal/[EMAIL PROTECTED];fs_path=bob.com' &echo()
/b
On Dec 9, 2008, at 1:52 PM, Michael Collins wrote:
> What SVN rev are you running? Also, could you do a SIP trace?
> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> Pastebin the output of that and we'll take it from th
What SVN rev are you running? Also, could you do a SIP trace?
TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
Pastebin the output of that and we'll take it from there.
-MC
On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
<[EMAIL PROTECTED]> wrote:
> Both:
>
> originate sofia/external/'[EMAIL PRO
Both:
originate sofia/external/'[EMAIL PROTECTED];fs_path=proxybeta.foo.net'
&echo()
originate sofia/external/[EMAIL PROTECTED];fs_path=proxybeta.foo.net
&echo()
produce the exact same result & log
:(
> * I think you need to '' the sofia uri /b
_
I think you need to '' the sofia uri
/b
On Dec 9, 2008, at 1:11 PM, Erick Johnson wrote:
> Looking at the logs the reason as to why it's been termintated isn't
> cleear
> to me. Any thoughts?
___
Freeswitch-users mailing list
Freeswitch-users@list
Thanks Brian,
I never did want to use a gateway - I was just lost on how to force FS
to use a proxy.
fs_path seems to be what I'm looking for. However now what I run my
originate command the channel gets terminated before FS even sends out
a packet. The api call completes with NORMAL_UNSPECIFIE
Can you clarify why you need a gateway? Is the far side going to
challenge us and request authentication credentials?
So you want us to not resolve the domain of the target at all in any
way? That kinda breaks the rules because you should always check the
NAPTR's and SRV and resolve to th
Hi Brian,
Thanks for the reply, but I still don't think that answers my original
question. I'm trying to get FS to act simply as a UAC in this
instance, what I want is for FS to proxy ALL outbound calls through my
proxy server at foo.com.
So when FS originates a call to [EMAIL PROTECTED] I want
see this is better.
That's why I asked you to be more specific about what you want because the
tiny back and
forth questions were not exposing your intent or needs at all. I answer
every email I can and when threads start to grow without getting to the
point i start to get frustrated.
Now that y
2008/12/9 Michael Jerris <[EMAIL PROTECTED]>
>
> On Dec 9, 2008, at 9:10 AM, Joe Bain wrote:
>
> Ok I have been testing more and I have reduced my problem to a pretty
> short and simple Lua script. I've posted it at
> http://pastebin.freeswitch.org/6373 and this gets called straight from the
> d
Woof!
It appears that FreeSWITCH writes
freeswitch.history
freeswitch.log
freeswitch.pid
freeswitch.xml.fsxml
to the -log directory.
Is there a way to put the files other than freeswitch.log into the -db
directory instead?
In my environment we archive and rotate everything in t
Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to
> catch all of the various variables (i.e. from XML_CDR) or otherwise can
> be a little trying (a row in your CDR database could be over 100 fields
> long!).
>
> The best option here is to
That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc)
That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc)
That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc)
Helmut,
I think Mike J was pointing out that spandsp needs libtiff and
libtiff-devel in order to compile, so you need to do that first and
then compile freeswitch.
-MC
On Tue, Dec 9, 2008 at 8:09 AM, Helmut Kuper <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi M
I'm running Gentoo Linux.
# uname -a
Linux 2.6.25-gentoo-r7 #1 SMP PREEMPT Sun Oct 5 01:51:24 PDT 2008 x86_64
Intel(R) Xeon(R) CPU X3320 @ 2.50GHz GenuineIntel GNU/Linux
/tmp is writable by everyone ...
# ls -ld /tmp
drwxrwxrwt 4 root root 4096 Dec 9 08:28 /tmp
ideas?
also, I assume the spoo
On Dec 9, 2008, at 11:09 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Michael,
>
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile
Hello,
This is just my 2 cents ... but my experience has been that trying to catch
all of the various variables (i.e. from XML_CDR) or otherwise can be a
little trying (a row in your CDR database could be over 100 fields long!).
The best option here is to catch the UUID's for the 2 call legs, capt
-BEGIN PGP SIGNED MESSAGE-
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Hi Michael,
don't know if you get me right: Everything is there, but obviously FS
makefile has to compile "libs/spandsp/src" before mod_fax (at least I
guess so). Currently the Makefile referred to libspandsp.la before it is
compiled.
regards
helmut
On Dec 9, 2008, at 6:52 AM, Jan Kubr wrote:
>> btw you can send
>>
>> call-command: hangup
>> hangup-cause: normal_clearing
>>
>> in place of
>> call-command: execute
>> execute-app-name: hangup
>> execute-app-arg: normal_clearing
>
> What is the difference this makes? Just curious because I've b
you don't want to be using the tone detect here, you want to be using
bind_meta, but without the meta key, which I don't think it can
actually do currently.
Mike
On Dec 9, 2008, at 8:31 AM, Frank @ Impact wrote:
We are actually trying to detect the called party pressing a key –
dtmf. In
On Dec 9, 2008, at 9:10 AM, Joe Bain wrote:
Ok I have been testing more and I have reduced my problem to a
pretty short and simple Lua script. I've posted it at http://pastebin.freeswitch.org/6373
and this gets called straight from the dialplan. From my experience
so far it only exits afte
Fixed in svn r10678. Thanks for the report.
Mike
On Dec 9, 2008, at 9:51 AM, Michael Collins wrote:
> Thanks again for the heads up. We'll check it out.
> -MC
>
> On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper
> <[EMAIL PROTECTED]> wrote:
>> I tried compile FS with mod_xml_ldap with trunk of ye
make sure you have libtiff and libtiff dev packages installed then re-
configure freeswitch
Mike
On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make'
> in
> FS trun
It is something we have been discussing as we need these stats to do
rtcp properly but we have not written any code to do so. It is
"somewhat" difficult. I would say it is on our minds but not on any
roadmap just yet.
MIke
On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:
> I'm curious
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of
Don't want the tone to be wrong here, but this makes no sense.
Carriers surcharge like this precisely to guard against call center,
predictive and other mass outbound calling scenarios.
It just doesn't make since, math wise, that individuals hanging up on
voice mail are going to significant
Thanks again for the heads up. We'll check it out.
-MC
On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> I tried compile FS with mod_xml_ldap with trunk of yesterday. During
> compiling it can't find
> http://sv
Which OS are you running?
-MC
On Tue, Dec 9, 2008 at 2:07 AM, Helmut Kuper <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make' in
> FS trunk directory led to an error saying that libspandsp
Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED]> wrote:
> "I already added 2 patches fo
First example is WRONG you don't dial via a gateway that way. If you
wish to dial [EMAIL PROTECTED] then try sofia/internal/[EMAIL PROTECTED] as
you don't require a gateway to call alice right?
/b
On Dec 8, 2008, at 7:14 PM, Erick Johnson wrote:
Here is the command that I'm trying to use
This is something that would make a great deal of the trouble shooting
easier. The Linksys SPA942's that we use have some stats available for
this, but it would be better to have it available centrally.
Scott
Jonathan Palley wrote:
I'm curious to start a discussion on being able to query a
ch
Hi There,
I'm trying to get freeswitch to originate all SIP calls through
an outbound proxy. When I use the originate API command to create a call
to a telephone number I see the SIP packets getting to my proxy just fine.
However if I originate a call to a SIP address then proxy server is
bypass
Ok I have been testing more and I have reduced my problem to a pretty short
and simple Lua script. I've posted it at
http://pastebin.freeswitch.org/6373 and
this gets called straight from the dialplan. From my experience so far it
only exits after a caller hangup about 1 in 10 times. Most of the ti
If you're running SELinux then you'll need to correct that on your
machine to allow FreeSWITCH to write to /tmp
/b
On Dec 9, 2008, at 2:43 AM, Gabriel Kuri wrote:
> I've been experimenting with mod_fax and discovered it doesn't
> appear to
> receive faxes unless freeswitch is running as root
id also love to get any info from the RTCP...
even have this in the XML CDR would be great..
would love to derive quality stats for calls based on RTCP
Jay
On Tue, Dec 9, 2008 at 2:37 PM, Jonathan Palley <[EMAIL PROTECTED]> wrote:
> I'm curious to start a discussion on being able to query a chan
How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)?
We have a carrier that penalizes us with a surcharge for short duration
calls (sound familiar?).
So when a FS caller (not a call center or predictive dialer) calls a
cell phone and gets a ring tone or calls
We are actually trying to detect the called party pressing a key - dtmf.
In band for ulaw. Rfc2833 for 729.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Jerris
Sent: Monday, December 08, 2008 12:57 PM
To: freeswitch-users@lists.freeswitch.org
> btw you can send
>
> call-command: hangup
> hangup-cause: normal_clearing
>
> in place of
> call-command: execute
> execute-app-name: hangup
> execute-app-arg: normal_clearing
What is the difference this makes? Just curious because I've been
using the latter as well.
> we just tested you chang
Updated and works great, thanks!
On Mon, Dec 8, 2008 at 6:18 PM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> i added a patch to index the variables on the
> SWITCH_EVENT_CHANNEL_EXECUTE_COMPLETE
> if you want to update
>
> otherwise you can use uuid_getvar to retrieve the variable
>
>
> On Mon,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I tried compile FS with mod_xml_ldap with trunk of yesterday. During
compiling it can't find
http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked
there and found that the filename on freeswitch.org side has changed to
http://s
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I tried to compile mod_fax today with trunk from yesterday. A 'make' in
FS trunk directory led to an error saying that libspandsp.la wasn't
found in libs/spandsp/src. So I had to configure and compile (make)
spandsp manually before compiling FS
2008/12/9 Ivan C Myrvold <[EMAIL PROTECTED]>
> Did you read carefully when asked to provide login and password? The login
> and password is there, don't use your own freeswitch login.
>
> Ivan
>
> Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>
> On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <[EMAIL PR
Did you read carefully when asked to provide login and password?
The login and password is there, don't use your own freeswitch login.
Ivan
Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm writing an IVR in Lua and am
On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm writing an IVR in Lua and am having problems dealing with hangups
> cleanly. Very often session:ready() reports true long after I have hung up
> and the hangup hook function I have set doesn't get called either. It
se
I've been experimenting with mod_fax and discovered it doesn't appear to
receive faxes unless freeswitch is running as root? it fails trying to
open the tiff file for writing (see the logs below). I'm using the
dialplan as prescribed in the wiki without any changes and the user the
freeswitch proce
"I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol
specific. So one would expect there to be an iax_hangup_disposi
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