The patch from the PortAudio site does get the library to build, but
it still fails with the same assertion when I try to play MOH. The
patch I'm talking about is this one:
http://www.portaudio.com/trac/changeset/1418
If the same build problem applies to other 64 bit systems, it might be
Hi,
While I appreciate that Fedora 12 is still only in beta. Because I
want to try FusionPBX and I've had no success getting pfSense to work
in a single NIC environment, and FusionPBX needs PHP 5.3, and Fedora
12 appears to be the first distro with PHP5.3...
Anyway, these are the steps that I've
Dear all,
I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE
event comes.
I executed the script for the 1st time and I got nothing.
When I executed the script for the 2nd time, it ended with Sedmentation
fault with core dumped.
I was unable to attach the core dump file
Any reason for not using uBuntu?
Install Freeswitch + FusionPBX on Ubuntu
step 1) add the fallowing lines to /etc/apt/ file.
deb
http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntuhardy
main
deb-src
Hi,
Does anyone know how to playback based on files from hadoop storage.
There is a libhdcp, and java api. Is there anyway to put together a sample
middle piece to move files from hadoop to freeswitch using memory space, so
there is no disk I/O?
Any feedback or suggestion will be greatly
Hello,
When you post something on pastebin, please post the link to your post so
everyone can find it, what is the link to it?
Have a nice day :)
2009/11/7 Dave Stevenson steve...@primrosebank.net
Hi Michael,
thanks for the reply. I think that I have got to the bottom of how to allow
Hi Ed,
I installed Jaunty ( I don't have Hardy to hand)
rather than /etc/apt, I presume you mean /etc/apt/sources.list
after a sudo apt-get update I did a sudo apt-get install
freeswitch I'm not sure what you meant by deps by your step 3
I then edited /etc/defaults/freeswitch and set false to
Milena,
thanks a lot for the reply - sorry, I'm new to this, but I'll remember that for
next time.
Actually, I found my way to the IRC site and the helpful chaps there got to the
bottom of my problem.
I had made an error copying the dialplan data from the SPA3102 FreeSwitch
HowTo
On Fri, Nov 6, 2009 at 7:59 AM, Ujjval Karihaloo
ujj...@simplesignal.com wrote:
Any examples I can refer to for this?
not that i know of
Like for Channel vars and execute_application calls? Does this all need to be
doen in dialplan.public.xml or also in other config files?
most can be
Dear Freeswitch Users,
I am looking for a SIP Provider who can provide a DID with unlimited
channels. Currently I am using junction networks but they have a high
2.9c/minute charge. I am looking for someone who has a flat rate for X
minutes.
Any advise would be much appreciated.
Thanks.
This is for outbound calls, calling party name. The OP is talking
about called party name, which is the neat feature of being able to
update the display of the calling user with the name of the called
user (instead of just displaying their numeric extension for the
duration of the call).
On Sun,
Beware of anyone that claims to offer unlimited channels. We're
still fundamentally a TDM world and there is no such thing as
unlimited. Depending on what you are looking for there are probably
plenty of providers with a high enough limit to satisfy your actual
needs. I just frown upon anyone
Have you tried Bandwidth.com or iCall?
/b
On Nov 9, 2009, at 8:44 AM, Shameem Shiek wrote:
Dear Freeswitch Users,
I am looking for a SIP Provider who can provide a DID with
unlimited channels. Currently I am using junction networks but they
have a high 2.9c/minute charge. I am
I agree there is no such thing as unlimited. The three ways most SIP
providers will structure pricing is 1) per minute (ie $0.02/minute),
2) per channel (ie $15/month) or 3) unlimited with a channel limit
(ie $7/month for any amount of minutes but after two simultaneous
channels its ring
If the patch is not received today it will not make it into 1.0.5
On Mon, Nov 9, 2009 at 9:30 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
This is for outbound calls, calling party name. The OP is talking
about called party name, which is the neat feature of being able to
Hi Ed,
I've just finished installing Hardy, and following the same steps
again, freeswitch is not running.
Any suggestions ?
On Mon, Nov 9, 2009 at 1:48 PM, shouldbe q931
shouldbeq...@googlemail.com wrote:
Hi Ed,
I installed Jaunty ( I don't have Hardy to hand)
rather than /etc/apt, I
You can check the numbers of arguments passed with argc, and access
them via argv[0], argv[1], etc.
Its hinted at on the main Javascript wiki page, and also detailed in
the FAQ.
http://wiki.freeswitch.org/wiki/Javascript_FAQ
On Nov 8, 2009, at 10:34 AM, god.nirvana wrote:
hi all:
1) install gdb
2) run support_d/fscore_db in the tree from the working directory of the
core.
3) if you are not on svn trunk, make current and start over.
On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:
Dear all,
I did the below code, to callback a function
maybe we should write a new audio abstraction lib =D
On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher br...@nani.ca wrote:
The patch from the PortAudio site does get the library to build, but
it still fails with the same assertion when I try to play MOH. The
patch I'm talking about is this
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
On Mon, Nov 9, 2009 at 9:01 AM, Fred-145 codecompl...@free.fr wrote:
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?),
I am looking on advice on how to set up a simple office PBX, 20 phones
and 4 outside lines.with 2 or 3 operator phones and the rest will be
extensions.
4 spa3000's to handle the outside lines.
2-3 polycom 601 phones with expansion modules (Operator phones)
18 polycom 330 or other phones
Dear All,
I couldn't find much information about how to monitor Freeswitch via SNMP
like how many calls/legs I have, how many CAPs, and etc. One of the thing I
do currently is to make simple bash script which in general runs fs_cli -x
'show calls count' or some other command and call that
While I'm very happy to hear this, the wiki has in more than one place
suggestions to install multiple sound and moh 'sets'...
On Mon, Nov 9, 2009 at 5:34 PM, Michael Collins m...@freeswitch.org wrote:
On Mon, Nov 9, 2009 at 9:01 AM, Fred-145 codecompl...@free.fr wrote:
Hello
I
On Mon, Nov 9, 2009 at 9:48 AM, shouldbe q931
shouldbeq...@googlemail.comwrote:
While I'm very happy to hear this, the wiki has in more than one place
suggestions to install multiple sound and moh 'sets'...
Link(s) please? I'll take care of the wiki.
-MC
I was sure I'd seen more, but
http://wiki.freeswitch.org/wiki/Installation_Guide search for There
are also higher bitrate sounds available for download and installation
with:
Cheers
Arne
On Mon, Nov 9, 2009 at 6:13 PM, Michael Collins m...@freeswitch.org wrote:
On Mon, Nov 9, 2009 at 9:48
On Mon, Nov 9, 2009 at 10:28 AM, shouldbe q931
shouldbeq...@googlemail.comwrote:
I was sure I'd seen more, but
http://wiki.freeswitch.org/wiki/Installation_Guide search for There
are also higher bitrate sounds available for download and installation
with:
Cheers
Arne
Thanks! I'll clean
2009/11/9 Dimitar Dechev ddec...@nutel.cc
Dear All,
I couldn’t find much information about how to monitor Freeswitch via SNMP
like how many calls/legs I have, how many CAPs, and etc. One of the thing I
do currently is to make simple bash script which in general runs “fs_cli -x
‘show
mercutioviz wrote:
I recommend you just do this: make cd-sounds-install make
cd-moh-install
Will do. Thanks.
--
View this message in context:
http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26271417.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
Or write one for Mac specifically since PA is fine for all the rest (I
think)?
JM
On Mon, Nov 9, 2009 at 2:50 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
maybe we should write a new audio abstraction lib =D
On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher br...@nani.ca wrote:
I have Siemens A58IP and Snom M3. Both work very well with pros and cons.
Nonetheless, both lack HD
JM
On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson steve...@primrosebank.netwrote:
Hi,
has anyone any good results to share with using cordless phones for VOIP
with FreeSwitch ?
I
I am also curious whether you can recommend how I can get the info if I want to
see concurrent calls by account code. Let's say if I am running FS as SBC and
I want to monitor concurrent calls per customer. I've looked at the HEARTBEAT,
but it only gives me overall session-count.
How safe is
Hi,
has anyone any good results to share with using cordless phones for VOIP with
FreeSwitch ?
I have seen a few around that appear to operate with wireless networks and make
SIP connections to VOIP PBXs.
I have seen various models from Engenius, Prestige, DORO and Siemens as well as
Snom.
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:
I just want to clarify the status of PortAudio on 64 bit architectures.
There is a compile-time problem in pa_dither.c (and .h) that comes
from the code not being 64 bit ready. This problem has been patched
cleanly here:
http://www.portaudio.com/trac/changeset/1418
I think this patch
On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote:
Hi,
Does anyone know how to playback based on files from hadoop storage.
There is a libhdcp, and java api. Is there anyway to put together a sample
middle piece to move files from hadoop to freeswitch using memory space, so
Joao,
thanks for the note. The Snom M3 is one of the ones that I was looking at - I
would be interested in the Pro's Cons ?
Interesting about the HD, but do you notice the difference and find that you're
disappointed with the quality of their sounds ?
regards
Dave
- Original Message
On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote:
Joao,
thanks for the note. The Snom M3 is one of the ones that I was
looking at - I would be interested in the Pro's Cons ?
RU
Interesting about the HD, but do you notice the difference and find
Hi,
thanks Brian, that's interesting. I had a comment off list which suggested
the same thing.
It did not quite fit with my aspiration for an all VOIP solution, but it sounds
like the technology is not quite there yet for hands-free.
That's great feedback before I spend some cash on a
Use mod limit to do this. You can choose to use it in count only mode
if you want (no limit).
On Mon, Nov 9, 2009 at 11:05 AM, DJB djbin...@yahoo.com wrote:
I am also curious whether you can recommend how I can get the info if I want
to see concurrent calls by account code. Let's say if I am
I agree about the M3. I have the handset and it is not ergonomic at all.
I also have a Siemens A580-IP. It does do G722 but has a few bugs
related to G722 that I normally run it with G711 only. There is a
quality difference between G722 and G711 when talking among the A580
handsets or the
I **think** that the following will match any three character strings from 1xx
to 399
I want to exclude 100 though, can anyone help me with the required RegEx please
?
^([1-3][0-9][0-9])$
I could (I think) do
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
But it does not feel elegant - is there a
On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote:
Get an ATA with a Dect handset it works much better... the Snom M3 and
the Aastra are one in the same and they both do not live up to the
quality or usability requirements.
That said, they are better than what else is around.
I'd call them
On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson
steve...@primrosebank.netwrote:
I **think** that the following will match any three character strings
from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx
please ?
^([1-3][0-9][0-9])$
I could (I think)
Would something like this work for you?
extension name=some-extension
condition field=destination_number expression=^100$
!-- do something --
/condition
/extension
extension name=another-extension
condition field=destination_number expression=^([1-9]\d{2})$
!-- do something else --
Hi Stephen,
thanks for the reply.
I'm not sure , does the code below handle all number from 101 to 399 ?
It would rely on the 100 code being picked up by the dialplan before the other
extensions were processed so the order of the code in the dialplan is
significant. Is that how people
Thanks Michael,
but I want to exclude 100 ?
regards
Dave
- Original Message -
From: Michael Collins
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, November 09, 2009 8:38 PM
Subject: Re: [Freeswitch-users] RegEx Help
On Mon, Nov 9, 2009 at 12:22 PM, Dave
On Mon, Nov 09, Michael Collins wrote:
On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson
steve...@primrosebank.netwrote:
I **think** that the following will match any three character strings
from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx
Hi Rupa,
thanks for the tip. I've had a look for the A580-PI - as you say, quite
inexpensive and probably worth taking a chance on one.
regards
Dave
- Original Message -
From: Rupa Schomaker r...@rupa.com
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, November 09, 2009 8:23
Hi Frank
Yup ! That's what I mean :-)
thanks a lot,
regards
Dave
- Original Message -
From: Frank Carmickle fr...@carmickle.com
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, November 09, 2009 9:03 PM
Subject: Re: [Freeswitch-users] RegEx Help
On Mon, Nov 09, Michael
Dave Stevenson steve...@primrosebank.net said:
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
Another possibility.
^(1(0[1-9]|[1-9]\d)|[2-3]\d{2})
--
Russell Mosemann
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news
On Mon, Nov 9, 2009 at 11:50 AM, Fede federico.om...@gmail.com wrote:
Hi!
I'm trying the Doodle web SIP phone but for some reason I'm unable to
register to my FreeSWITCH server. I've tried with other servers and it works
ok.
Did someone tried this web phone with FreeSWITCH? Any tips why it
Dave,
I think extensions are processed in order although I can't quickly find any
documentation that says this, why don't you try it and see, it would take
only a moment to find out for sure.
--Stephen
On Mon, Nov 9, 2009 at 1:03 PM, Frank Carmickle fr...@carmickle.com wrote:
On Mon, Nov 09,
On Mon, Nov 9, 2009 at 1:10 PM, russell.mosem...@cune.org wrote:
Dave Stevenson steve...@primrosebank.net said:
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
Another possibility.
^(1(0[1-9]|[1-9]\d)|[2-3]\d{2})
Yep this is the one. I'm sorry I didn't read the OP correctly the first
time. Skipping
Hi Michael!
Thank you for your quicky answer.
I'm using FreeSWITCH 1.0.5 pre5. The debug log from the command line plus
the SIP trace are at: http://pastebin.freeswitch.org/11043
The Doddle web phone is at: http://www.doddlephone.com
You can test this account at my FreeSWITCH server at:
I just tried the webphone with my freeswitch server and it worked fine,
making a call to my echo test w/o any issues...so it's probably a
configuration issue with freeswitch.
--matt
http://www.hellohunter.com
On Tue, Nov 10, 2009 at 4:15 AM, Michael Collins m...@freeswitch.org wrote:
On Mon,
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
___
Hi Guys,
OK, with the RegEx help that you gave me, I have separated out the processing
of extension 100 from 101 to 399 as I wanted.
I have created a group (100) which contains a number of phones - 101 to 105 at
the moment.
When the PSTN line rings, I want all the extensions in the group to
Thanks - pretty unambiguous reply !
I won't go down that route then :-)
- Original Message -
From: Joseph L. Casale jcas...@activenetwerx.com
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, November 09, 2009 10:46 PM
Subject: Re: [Freeswitch-users] Cordless VOIP Phones
The
asstra has one issue where if you look at them wrong they start telling the
server that the media ip is 0.0.0.0 which we have never identified but they
indeed seem to work better than snom m3
On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale
jcas...@activenetwerx.comwrote:
The Snom M3 is one of
Beat me with a dead cat all you want but I rather the snom m3 than the
Siemens A580IP Siemens has very low volume which makes its call quality
suck despite of being ergonomic and all...
That gigaset application sucks and the base station is slow as hell... Maybe
I have a bad unit?
The snom
See comment inline
On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson steve...@primrosebank.netwrote:
Hi Guys,
OK, with the RegEx help that you gave me, I have separated out the
processing of extension 100 from 101 to 399 as I wanted.
I have created a group (100) which contains a number of
Hi,
From Freeswitch there is continuously Request: Notify (Messages-waiting)
requests are comming, i didnt subscribe from Freeswith and pjsip(ua).
any body know how to stop those requests from Freeswitch.
Thanks--
Srinivasula Reddy K
___
param name=send-message-query-on-register value=false/
Add that to your sofia profile.
You must be new to SIP, you will soon learn that almost every SIP device
just stupidly expects you to send this and never does it the correct way by
subscribing to it which is why this option is the default.
Well,
I thought it was fixed - it is more or less working, with one more stumbling
block.
I have just posted a dump to the pastebin - from Dave (stevendt)
The voice mail works - but too well.
If the call is answered by a someone at this end - everything is fine until the
user hangs up, then
I have asked you before to please not cross post to both mailing
lists. Please refrain from this in the future.
Mike
On Nov 9, 2009, at 6:36 PM, srinivasula reddy wrote:
Hi,
From Freeswitch there is continuously Request: Notify (Messages-
waiting) requests are comming, i didnt
You set both hangup_after_bridge and continue_on_fail after you already
called bridge.
Try setting it *before*
Seems to be a running theme here that things will be parsed in a linear
fashion that you may want to take note of.
On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson
Oops, you've got some lines that are in the wrong place:
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
Those lines need to come prior to the bridge call or they'll never be
applied. :)
-MC
On Mon, Nov 9, 2009 at 4:18 PM, Dave
Thanks Anthony
that did the trick !
Excuse my ignorance - this is all new to me . . .
It would help if I knew what I was doing, as I commented below, I copied the
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
from the code for the
Dear Freeswitch users,
I am building an app where the extensions map to external callers and there
are no registered users. For example, the extension 1001 would map to an
external number. In that case, does it make sense to use the Mod voicemail
or should I build a voicemail solution using
OK, I may have solved this mystery, if I use application=answer and answer the
call before the IVR which then flows into the Conference app, DTMF works from
the ATT phone..
So, if you face issues with Conferencing/IVR, answer the call before you invoke
those apps...
Problem I have now is that
Hello All
I am trying to configure freeswitch so that it sends outgoing calls to the
PSTN through voicepulse
My configuration is as follows.
I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml
include
!-- West Coast gateways --
gateway name=voicepulse
param name=username
Paul Thirumalai paul.thiruma...@gmail.com wrote:
I am really new to VOIP and having a hard time with this. I am really not
sure how to proceed. Any help would be really appreciated.
First, turn on debug logging (in fs_cli, it's /log debug) to obtain more
information.
The proxy variables in
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