mercutioviz wrote:
Additionally, turn on debugging on the console and capture that output. If
you use fs_cli it has debug output turned on by default.
Thanks for the tip. I launched fs_cli, typed sofia profile internal
siptrace on, and then made a call from XLite to the GS phone, with the
I guess I can limit the amount of debug data in the CLI by choosing the right
debug level:
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
What is the recommended way to debug SIP connections like I'm having?
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FWIW, I downloaded and compiled the latest trunk (16041), and am still having
this issue.
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More information: I can dial the default extensions like just fine. It's
only when I call any of the IP phones (1001,1002,1003) that the call is
immediately forwarded to the callee's voice-mail when the phone goes off the
hook.
To only keep the SIP messages in the fs_cli screen, typing
Hello
I'm running 1.0.trunk (15841) on Linux CentOS with a the default settings.
After succesfully connecting a Windows PC running XLite (EyeBeam, really)
and a GrandStream IP phone to Freeswitch, I try to make calls, but am having
the following issues:
1. When calling XLite from GS, XLite
I found the cause for #2: The GS phone was still configured to use NAT, even
though both XLite and GS are located in the same, private LAN. Unchecking
this on the GS phone solved the issue.
But I'm still having issue #1, regardless of which phone is calling or being
called: When the phone
I don't have access to a remote computer from which I could log on and run
nmap.
I'll see if I can get a shell access somewhere. Thank you.
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Hello
I'm no wget expert, and figured I should ask here first: I'd like to
download the whole wiki using wget for off-line reading.
Using the following didn't work:
wget -m -np http://wiki.freeswitch.org/wiki/Main_Page
If I move the wiki/ directory to the root directory of my web server, and
I only tried once. Maybe someone used to wget could generate a PDF in case
people need an offline copy?
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Hello
Does someone know of a free service on the web that can check whether the
UDP ports on my firewall are open after Freeswitch is up and running?
ShieldsUp only scans TCP ports.
Thank you.
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Are you typing up something for posterity's sake? If so let me know. I'll be
happy to proof-read the finished product and offer suggestions.
Thanks much for the clarification. I'm used to writing short documentation
when I learn a new tool, so 1) it helps me understand how it works, 2) I can
Hello
I'm about to test Freeswitch on Windows, and would like to make sure I get
it right:
http://files.freeswitch.org/windows_installer/
contains freeswitch-1.0.4.exe from 03-Sep-2009 and freeswitch.exe from
07-Dec-2009. Am I correct in understanding that I should run the former, and
then
I'll answer my own question ;-)
Freeswitch.exe = svn 15826, ie. a temporary installer until 1.0.5 is
available.
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I have a suggestion to make for the Windows installer: Make installing
FreePBX/FusionPBX an option, as it adds 100MB although not everyone wants a
web GUI to manage Freeswitch.
My .15€
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Hello
Since Freeswitch is also available for Windows (and Mac, but I don't
anything about Macintosh), I'd like some feedback from users who routinely
run Freeswitch on that OS.
Is it stable enough to be used in production to handle a single analog line
(ie. SOHO use), or should I warn customers
The 3102 has quite a lot of settings you can play with. Here are two useful
documents:
SPA-3102 Simplified Users Guide Version 1.1a
http://www.jmgtechnology.com.au/spa_3102_guide.pdf
Linksys ATA Administrator Guide 3.2.pdf
http://www.inphonex.com/download/spa8000-ag.pdf
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Hello
I'm a bit confused at the difference between those two concepts. Contexts
are created in the /dialplan, and are refered to by items in /SIP_profiles
and extensions in /directory.
What purpose do contexts and profiles play?
Thank you.
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Thanks Anthony for the tip.
Would you say this is a correct representation of things?
Contexts are a set of extensions in conf/dialplan/ (eg. default, public,
etc.)
Extensions are configured through files in conf/directory/. Each extension
maps to a context (variable name=user_context
Michael Jerris wrote:
we also support natpmp and static ip setting.
What is static ip setting? Telling FS what the public IP is? If that's
what it is, what about the UDP ports that must be open to allow incoming
connections?
So, in the case where the FS server is located in a private network,
One last question: Does someone know of a utility for Windows that can check
that a NAT router supports either UPnP or NAT-PMP? I guess it's no big deal
to write a small diagnostic by connecting to free firewall checkers to see
if the relevant ports are open, but if it's already available...
No publisher, although uploading and selling books (deadtree or online) is
easy with companies like www.lulu.com
I was just thinking of some way to learn FS gradually and effectively. The
frequent problem with wiki's, is that the quality of articles is uneven and
they don't have a good layout.
Frank Carmickle wrote:
A board with an atom 330 on it would probably do the trick for you. There
are a few made by Intel and Supermicro that look pretty nice. There were
some other people on the list looking to use them. Maybe we can get a
report from someone.
Intel came up with the
Mark Campbell-Smith wrote:
I use a dectop by Data Evolution... Its cheap at ~$100. I have it
running debian lenny and FS... works well for me.
Thanks for the tip, although this type of box doesn't have a PCI slot, so
the only way to connect FS to the PSTN is through a VoIP provider (or a
Hello
I'm going through the various XML files, and noticed this first line in
vars.xml.
X-PRE-PROCESS cmd=set data=default_password=1234/
What is this password used for?
Thank you
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Hello
I wanted to check if my ADSL modem worked with STUN, so I left its UPNP
activity option unchecked, ran FreeSwitch, and used eg. Shields Up
(www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened...
which SU says no.
Does it mean that...
- by default, FS doesn't use STUN
- or
Ah, makes sense:
conf/directory/default/1000.xml:
user id=1000
params
Thanks for the tip.
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Thanks for the clarification. So it's either UPnP or STUN/port-mapping.
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Michael Jerris wrote:
Our plan for 1.0.5 is that we will also have rpm and deb packages for many
distros on our own repo. Stay tuned. This has been another major reason
for the delay in 1.0.5.
Great news. I also prefer to use packages whenever possible, so as to know
what software is
Nik Middleton wrote:
I cannot imagine doing what I'm using FS for, with any other product. Yes
it's frustrating at times, but this is largely down to a lack
documentation/samples.
Speaking of which... would this layout be good for a book on Freeswitch?
Preface
1. VoIP, Freeswitch, FS vs.
Hello
I'd like to install OpenZAP so I can use a TDM card with Freeswitch, but I'm
getting a software error althought the TDM card seems detected (lspci -v
OK). Dahdi was successfully compiled from source code.
Is it OK to just install Dahdi 2.2.0 without Asterisk before going ahead
with
Thanks Russel for the tip. After more googling, I ended up figuring that
/etc/dahdi/modules had to contain the list of drivers to load.
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
Russell.Mosemann wrote:
You only need one of the modules above, if you have one card. I don't see
a dahdi module listed in the file here.
Yup, turns out wcfxo is needed for the X10xP card, while wctdm is needed for
Digium cards. As for dahdi, maybe wcfxo/wctdm loads the dahdi module
Michael Jerris wrote:
with a client that does not support stun or at least rfc 3581 the results
are much more sketchy and require more hacks on the server side, but with
enough effort can almost always be made to work.
Thanks Mike for the feedback. If a user has a problem using my FS server,
Hello
In a thread back in March, I read that support for IAX in FreeSwitch is a
bit of kludge and since there's not much demand for it, chances are it won't
improve in the foreseeable future.
So I'd like some feedback from users who routinely connect to a FreeSwitch
server from various venues,
Mitch Capper wrote:
I did something like this recently.
Thanks for the feedback. I'll see how Linux can be made to send stuff to a
USB display.
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mercutioviz wrote:
I believe that French and Spanish sounds are in the works by the
community.
The only other sounds I'm aware of are the Russian ones.
Thanks for the tip.
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Russell.Mosemann wrote:
Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel
is not being developed anymore.
Thanks for the links. Turns out this card seems incompatible with the
motherboard I have, so I'll concentrate on the Linksys 3102 instead.
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Hello
I was wondering if someone had succesfully configured FS to display caller
ID on a LED like this?
http://usb.brando.com/prod_detail.php?prod_id=00575
That would be a nice alternative to displaying CID information on the user's
PC screen when users need to see who's calling where they're
... or alternatively, on one of those USB digital picture frames?
www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B87BHC
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Are non-English sound files available in the SVN version of the code?
I just tried installing the French sound files, but got an error:
Unknown target cd-sounds-fr-install
Unknown target cd-moh-fr-install
make[1]: *** [cd-sounds-fr-install] Error 1
make: *** [cd-sounds-fr-install] Error 2
Hello
I successfully installed FreeSwitch from SVN, and am now prompted to install
the sound files. Am I correct in understanding that sounds are POTS-grade
files (8KHz?) while cd-sounds are closer to VoIP-grade (16KHz?), and
hd-sounds and uhd-sounds are for Skype-grade sound files?
In that
mercutioviz wrote:
I recommend you just do this: make cd-sounds-install make
cd-moh-install
Will do. Thanks.
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Hello
For those of you running FS on CentOS (5.4) who compiled FS from SVN, I'd
like to make sure I'm doing it right to have FS start automatically at
boot-time:
1. cp /usr/src/freeswitch/build/freeswitch.init.redhat
/etc/init.d/freeswitch
2. vi /etc/init.d/freeswitch:
Hello
As an alternative to more expensive alternatives like OpenVox or Sangoma,
I'd like to order an X100P clone from www.x100p.com for use in France.
According to a PDF on the site, the reason this card gets bad reviews is
that the Silicon labs Si3012/Si3035 DAA chip used in the original
Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
the registration process, while a PBX offers services like ringing phones,
setting up calls, handling voice mail, etc.
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Thanks for the link. But then, since OpenSIPs doesn't run on Windows, I'll
stick to FreeSwitch :)
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Thanks Mike for the link. I'll investigate more whether running FS on
ARM-based devices is a good idea.
For those interested, another thread on the subject:
http://www.nabble.com/Freeswitch-vs.-Asterisk-speed-on-ARM-td25086585.html
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Hello
I was reading an article the other day about Freeswitch in which the author
used the terms SIP proxy and PBX. Is an SIP proxy just a way for SIP
clients to register and connect to each other, while the PBX part adds
services like voice mail, etc.?
Thank you.
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Hello
I don't have the technical expertise to tell, so here goes: Unless mistaken,
Freeswitch is written in C and/or C++, so I guess it's not linked to the x86
instruction set.
Is so, would it be a lot of work of porting it so it runs on ARM processors?
It'd be cool because it could run on tiny
Thanks for the link to PORTech's MV-370. For those interested, it can be had
for a retail price of £150/€165 (before VAT).
www.discountphonesystems.co.uk/acatalog/MV-370.html
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Or as a more affordable solution... is it possible to connect an entry-level
GSM phone to a PC running Freeswitch and use this as a poor man's gateway?
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Thanks Tihomir for the link.
From what I read, it appears that EdgePBX's FX02G is a full-fledged Asterisk
server with a GSM module and an FXS module. Did you reflash its NAND to run
Freeswitch?
At $300, I guess customers will rather take a subscription with a VoIP
provided and use their GSM
Hello
I'm selling a basic solution for SOHO customers (FS is installed on their
work computer running Windows or Macs) to handle an analog phone line.
When they're on the road, in addition or instead of getting a notification
by e-mail when someone calls their office, some users might want to
jfenton wrote:
If you look on there there's a clear acrylic case for the BeagleBoard
- I haven't seen any others available.
That's often the problem with appliances: Once the CPU/modo, RAM, SDD/HD,
PSU, case, and a PCI FXO board are computed, we end up with something as
expensive as a
EdPimentl wrote:
Back in April I posted these links on the list, in regards to a similar
question
Thanks Ed, but the problem with all those, is:
- they typically have so little RAM/Flash RAM that they can't run a regular
Linux distro, which means that we're stuck with whatever software is
Hello
Atcom's IP appliance is nice but it uses a BlackFish CPU, which is not
supported by FreeSwitch yet.
What FreeSwitch-appliance do you is the cheapest, most compact out there? If
possible, I'd rather a device that has room for an SDD or 2.5 HD so as to
be able to use a standard *nix distro.
Steve Underwood wrote:
Nobody has yet adapted Freeswitch for the Blackfin, and they probably
won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux,
which is a cut down Linux for machines of this type. It is quite
troublesome to get memory management to behave sanely on a
Hadley Rich wrote:
The issue you are going to have is that the IP0x are based on the
blackfin processor which as far as I'm aware FreeSWITCH doesn't compile on
yet. Last I heard there's an issue with APR.
Too bad. Thanks for the tip.
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Hello
Atcom's IP01 unit (www.atcom.cn) can be expanded to have 128MB RAM and 1GB
NAND flash. Before I go ahead and check, would someone know if a minimal
Linux + LAMP server* + Freeswitch can run OK with this amount of memory?
Thank you.
* I think I'll trade MySQL with Firebird, to avoid
Hello
I was wondering if someone had successfully ran a PBX on those tiny, $100
devices that run Linux?
www.pogoplug.com
www.plugcomputer.org
I'm thinking of hooking up a Sangoma U100
(http://wiki.sangoma.com/sangoma-wanpipe-usbfxo) to the USB port so it can
handle a couple of analog lines.
Hello
I browsed the wiki + archives of this list, but didn't find an article on
how to add a TDM card to Freeswitch. If I've overlooked it, thank you for
pointing me to it.
I have a one-FXO module OpenVox PCI card. It's correctly detected by Linux
CentOS 5:
# lspci -v
03:00.0
mercutioviz wrote:
If I understand correctly the OpenVox card is a zaptel clone, which means
you'd need to download the drivers, install them and then do the OpenZAP
install. See the OpenZAP page on the wiki.
Thanks for the tip. I'll give it a shot. In the mean time, I'm interested in
Antonio Gallo wrote:
Alix cases are like 6/9 € from their shop site. I think its easy to find
someone who work with aluminium that can make for you custom boxes for
like like 6/20 € at pcs
Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2)
allow for a PCI slot,
Mitch Capper wrote:
You may want to look at the Intel Atom combo machines you can get a 1.6
ghz machine probably for around $100-150 USD in a very small form factor
and very powerful.
Thanks for the tip. Any link where I could check this out?
The cheapest PC's I find are 230€ (Asus'
Thanks Antonio for the links on Acrosser and PCEngines. It seems like PCE's
alix1d is a good solution, provided 256MB is enough to hold Linux +
Freeswitch + some tiny LAMP stack. Still, it looks like an Atom-included
mobo like those from Asus or Gigabyte would be cheaper. The biggest issue is
Thanks guys for the links on CF-to-IDE adaptors.
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EdPimentl wrote:
Here is a list of the resources posted on this thread
After giving it more thoughts, I got to the conclusion that I'd rather a
stand-alone miniPC that can take a PCI card with a riser, instead of a
really tiny box that relies on an external box to connect to a PSTN line.
Thanks Brian. I'd like to make sure I finally got how dialplans, contexts,
domains, SIP profiles, and extensions in the directory work together:
Freeswitch supports different SIP profiles in conf/sip_profiles/
(internal.xml, external.xml, etc.), which are loaded through
Thanks for those pointers. There's several nice-looking devices in this
list:
http://www.linuxdevices.com/articles/AT2016997232.html
Ideally, I'd like a compact box that can fit a PCI card horizontally so I
don't have to provide a second, external device to connect the server to a
POTS line.
Thanks a lot for the pointers :)
At this point, alternatives $100 to a regular PC or some appliance like
the FritzBox or the Pika Asterisk appliance:
Planex MZK-W04NU $60
Sheeva Plug $99
(Deprecated) Linksys NSLU2 $50
Does someone know of other solutions that can run Linux/FreeBSD along with
David Knell wrote:
You might want to take a look at this:
http://www.amazon.com/IEEE802-11N-Wireless-Broadband-MZK-W04NU-Designed/dp/B000YDS0YG
Interesting :-) Thanks for the pointer.
I think it'd be very useful if the wiki included a list of devices that can
run Freeswitch in addition to
Fred-145 wrote:
I think it'd be very useful if the wiki included a list of devices that
can run Freeswitch in addition to a regular PC, especially those that cost
$150.
Here's a list of makers of basic routers per www.cheaprouter.us:
2Wire, 3Com, ADTRAN, Apple, Belkin, CNet, CP
Hadley Rich wrote:
That's what the wiki's for. If you create a login you can start a page
and start adding devices.
Yup, I'll start an article when I get enough information.
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Hello
For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and
simultaneous SIP calls) who'd rather not use a big, noisy PC to run
Freeswitch, I'd like to offer an alternative that has the following
features:
- not old hardware sold on eBay, ie. it must be up-to-date
xbipin wrote:
i was just thinking of throwing away my spa3102 and setting my windows
machine with FS on it to conenct to the telephone line for incomming
calls, something like a calling card gateway using a FXO card. the reason
being from the time i bought the spa3102, i haven't been able to
EdPimentl wrote:
This company may have what you are requesting:
Thanks. The SolidLogic Little Falls 2699 Mini-ITX @ $212 looks nice.
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Michael Jerris wrote:
It means the reason those directories where like that in the file were
because they are based on the locations the rpm puts the files.
Thanks again Mike. I went ahead, created a freeswitch user, chowned
/usr/local/freeswitch/ recursively, ran /etc/init.d/freeswitch,
Brian West-3 wrote:
The internal profile requires authentication. Every user on that profile
would use the user_context variable to override the profile context. Me
being paranoid when I wrote the configs I set the internal profile to
public just incase you misconfigure your system then
Michael Jerris wrote:
Those files in tree go with the rpm build with goes correctly to those
directories.
Thanks Mike, but I don't understand :-/
Do you mean that I should not bother compiling from SVN and download an RPM
instead (I didn't see a source listed under
Hello
I'm going through the various configuration files to figure out how they
work together, and noticed the following references: sofia.conf.xml -
sip_profiles/internal.xml - public context.
1. Am I right in understanding that a context is a file in the dialplan/
subdirectory?
2. Shouldn't
Hello
I'd like to make sure I understand this correctly: It seems like all the
settings found in autoload_configs/*.conf.xml are loaded, even if their
parent module is commented out in modules.conf.xml?
For instance:
-rw-r--r-- 1 root root 575 Apr 20 20:53 python.conf.xml
But..
Hello
I'm installing Freeswitch on a CentOS 5.3 test host. I noticed that the
freeswitch.init.redhat contains paths that don't match a stock CentOS
install.
All files are currently installed/owned by root: Should I go ahead, create a
freeswitch user/group, and chown everything under
mercutioviz wrote:
Anyway, the bottom line is that any XML file in autoload_configs/ will get
loaded into the big XML configuration file that gets stored in memory,
even if the corresponding module is not loaded from
modules.conf.xml.
So all configuration settings found in
Hello
I'm reading the examples, and found one on how to use a Sipura SPA-2000:
http://wiki.freeswitch.org/wiki/Sipura_STUN
I was wondering what the VIA messages mean, and whether I should enable
those when the unit is used behind a NAT router. I thought Freeswitch would
take care of NAT, but
Hello
I'm reading through the articles on the wiki, and have a basic question
about how Freeswitch handles the XML files: I noticed that conf/sip_profiles
contains both internal.xml, internal/, and internal/example.xml: Does it
mean that Freeswitch merges everything it finds in this location (in
Hello
I'd like to know how to set things up when using the following scenario:
- a VoIP gateway on the same LAN as the Freeswitch to handle incoming calls
from a POTS line
- a remote SIP phone somewhere on the Net
- the FS server and the remote SIP phone are both behind a NAT router
- the remote
Sorry, the URL was cut:
http://img14.imageshack.us/img14/2574/freeswitchlinksysnat.jpg
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mercutioviz wrote:
definitely stop by IRC and talk real-time with others who've dealt with
this
kind of thing.
Thanks for the tip, but since this firewalls are a frequent issue with SIP,
I think it'd be very useful if someone who has good experience with this
wrote an article in the FS
Diego Viola wrote:
FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy.
Thanks Diego. Based on this list features list, what does FS offer that
OpenSIPs doesn't?
http://www.opensips.org/index.php?n=Resources.Features
I don't know enough about VoIP etc. to be able to tell, but at first sight,
mercutioviz wrote:
Just following up... did you get these questions ironed out?
Not yet, but I do need to have a clear understanding about how to set things
up when NAT is involved, especially when remote SIP users are also behind a
NAT router, and especially if they can't make any change to
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