This is a total work in progress that has not even merged into tree. So it is
not known
to work or not work anywhere. Patches to correct issues are welcome.
Mike,
I took another look at this and don't really know enough about rpm building
to diagnose this. Frankly, the format of the latest
Am I correct in presuming that Freeswitch will answer a fax from a local zap
based user
just like it does from an FXO port connected to a POTS line? What I hope to
do here is
catch any call made from that extension (the zap based fax machine/user) and
push its
call into the fax module.
For the path in the dialplan I don't think we have any right now but
file a bug on jira and I can try to add them. As for something in the
script itself that is a bit more work but if anyone has a patch to
inject some vars into scripts like that it would be a nice addition.
Mike
Ok, signed up
So the spec from trunk says Soundfiles are moving into a separate spec
but I can't find this spec anywhere in svn?
Anyone know where it is?
Thanks!
jlc
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Working on it, moving the repos around to do this right...
http://jira.freeswitch.org/browse/FSBUILD-218
Mike
Thanks, Is this known to not work with non root builds? It errored out after
creating some
messy hierarchies with the actual variable calls, instead of their values?
Thanks!
jlc
Thanks, Is this known to not work with non root builds? It errored out after
creating some
messy hierarchies with the actual variable calls, instead of their values?
Actually,
I tried on a lab vm as root in the typical dirs. and got the same result:
...
`./us/callie/time/48000/hours.wav' -
Searching through the wiki for any indication as to what if any variables exist
for the install location in that I can leverage in a script.
Can anyone point me along, I can't seem to find anything. I want to place a
shell
script in /opt/freeswitch/scripts that needs a reference to a conf file
Trying to setup a new config in the pfSense 1.2.3 final package and when
I try to connect to the console I get an auth error?
# ./fs_cli -H 10.0.0.1
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Authentication Error]
I tried to search for docs to indicate where one might set the
Anyway I sent an email to Yawarra to ask them if the net5501 computer
http://www.yawarra.com.au/product.php?productCode=HW-NT55 is
compatible with the TDM400 cards.
It is, people have been doing this for a while w/ astlinux:
For those interested, here's how to compile and install Dahdi (which doesn't
need Asterisk at all, unlike some docs on the Net seem to imply due to
references to /etc/asterisk/*.conf):
I understand that Some Debian based distro's have Dahdi in their repo's making
it
simple, but not many know
Registrations:
=
=
As far as I can see, everything looks ok, except for the
2009-12-06 19:35:43.111477
In this case, the $1 will only contain whatever is in the parens in your
expression, i.e.
condition field=destination_number expression=^(\d+)$
What do you have for your expression?
-MC
Well, untested of course as I am busy with school:) But what I wrote up to try
at
Christmas (with your
Just remove the terminating '/' at the end of the second condition tag
condition field=destination_number expression=^(\d+)$
I tried to see based on examples if it was obvious to me why that should not be
there
but it didn't jump out:) Cuold you explain that please?
The gateway will arrive
To expand on what Russell said: XML always has a start and an end tag,
possibly with other stuff in between.
tag
... content ...
/tag
/snip
Ahh, so must all the actions be contained within at least one condition tag as
content, or could have I kept the
last / on the last condition and
Of course. Please read through the default configs and the getting started
guide and xml dialplan information on the wiki.
Mike
This is of interest to me as well, would that be something like this:
extension name=ext_100_vm
condition field=caller_id_number expression=^100$/
condition field=destination_number expression=^${caller_id_number}$
Of course:) Thank you!
jlc
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I need to make faxing easy for some very computer illiterate folk. I am using
an email
service and going to use procmail to print anything incoming automatically but
they cant
get the hang of scanning to an email app, so I am going to buy a Linksys PAP2T
as per the
wiki.
Since the setup will
Not sure where to start with this one, the outgoing leg to our sip
provider sounds perfectly fine but with the our M3's the incoming leg
is super choppy. Using twinkle on my laptop yields good results in
both directions so there must be an issue with just the snoms, their
firmware was very old but
Is there any way that each extension in the group can be notified that a
group Voice Mail is waiting to be picked up so that each phone shows the
message waiting indication ?
Wouldn't this be simply accomplished by setting the vicemail as box 100 for
each of the users (such as ext 1011xx)?
The Snom M3 is one of the ones that I was looking at - I would be interested
in the Pro's Cons ?
Worst POS I have ever used, from a sound quality to ergonomics pov, tech
support was as bad...
I have Aastra 480i CT's which work well.
jlc
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The internal.xml also has an ext-rtp-ip variable and in trying to
understand what this is for (my version of fs is 1) I noticed in trunks
conf file it is explained. So the available options that I have given
my setup is multihomed with a lan/wan setup where the wan interface is
dynamic would be a
auto-nat tries to use upnp/nat-pmp to figure it out... auto will just
put your IP in there.
The other values can be stun:host or an IP.
The docs in trunk show this now... its really simple to understand but
you should NEVER have to set that unless you have a nat scenario that
requires you to lie
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have through the cli?
Thanks,
jlc
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If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip
REMOVE them. If your multi homed then you'll need to set them.. we
don't listen on 0.0.0.0 you'll have to start a profile for each IP you
wish to listen on.
I am multihomed, and the wan nic is dynamic. Is there any way for me
Just use ${local_ip_v4} then.
and enable auto-restart on the sofia.conf.xml
Cool, it seems to always use the public ip, quite reliably.
That is what I am after (why), is there something in the code that
forces it to favor for example, non RFC 1918 addresses?
It works, I just want to understand
I am setting up fs in pfsense.
Following the multihomed tutorial (I also have a dedicated wan/lan int)
if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks
everything, but if I set X-PRE-PROCESS cmd=set data=domain=10.0.0.1/
in conf/vars.xml I now get audio working correctly,
It's failing to build the core library. There should be some warning
before it tried to build the modules in the log.
You'll have to bear with me, I am not sure exactly what part of the build
that is, I see this in the log:
+ FreeSWITCH Build Complete ---+
+ FreeSWITCH has been
Scroll up and look for the real error.
All I see are these:
*** Warning: Linking the shared library libfreeswitch.la against the
*** static library libs/libedit/src/.libs/libedit.a is not portable!
** Integer sample type enabled **
*** The gtk-config script installed by GTK could not be
I have recently updated the RPM spec for FreeSWITCH to use the latest
SVN release, including some unspecified files for the newest mods (with
nibblebil, unimrcp, etc, just like your build output is bitching about).
Follow these steps to build it:
Raul,
Appreciate this, it worked. I am re-running
I attempted to build rpm's from the included spec file using a non-root user
build environment. The steps I used are as follows:
1. Check build deps @
http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS
2. Pulled a copy of trunk in the SOURCES directory tar/bzip2 it as expected
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