So I did some tests and still I can not see CLIP on a phone connected to
an FXS port. Whether the call is bridged from SIP UA or from an incoming
call on FXO port does not change anything. Whether the parameter
enable-caller-id=true is present or not in openzap.conf.xml does not change
anything t
Hello
I am experimenting with FS and would like to know how to connect two
independent servers with user on one beinng able to call users on the
other. Do I set each server to be the gateway of the corresponding one ?
Pardon me if this has already benn dealt with. My search has drawn a blan
Hello
We also faced the similar issue. Actually it is caused bacause hold on music
files are missing. either you save all the music files or configure your
dialplan accordingly.
Sharad , Coral Telecom, India
- Original Message -
From:
To:
Sent: Wednesday, December 02, 2009 3:24 PM
Hello,
I created an extension in my dialplan so that when an incoming
call arrives, it rings a group of lines and then fallback to the voicemail
if no line is answered.
I wanted then that when voicemail starts, the
calling party could dial some numbers to fetch the voicemail. I used
bind_meta
Hi all,
I'm evaluating FS for our organization.
I must fulfill the following requirements:
1. Call recording: All (or selected) calls to the secretary must be
recorded.
2. Call barge in: Assume that two subscribers are talking to each other.
Secretary makes "emergency" (for example, an exten
On Wed, Dec 02, Fran??ois Legal wrote:
>
>
> Hello,
>
> I created an extension in my dialplan so that when an incoming
> call arrives, it rings a group of lines and then fallback to the voicemail
> if no line is answered.
>
> I wanted then that when voicemail starts, the
> calling party could
On Wed, Dec 02, Otis wrote:
> Hello
>
> I am experimenting with FS and would like to know how to connect two
> independent servers with user on one beinng able to call users on the
> other. Do I set each server to be the gateway of the corresponding one ?
You can if you need them to authent
What would be the best way of making originate() run call through a dial plan
(compared to directly going to a specified VOIP gateway). Would it be
loopbacks, i.e. smth like this?
/opt/freeswitch/bin/fs_cli -x "originate
{ignore_early_media=true,origination_caller_id_number=xx}loopback/yy
Hello,
We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI
there is a Nortel with Q.Sig. After a lot of configuration trials I've
managed to set it to send back the connected name over the SIP (i.e. when a
call goes from SIP to PRI, the PRI sends back the connected name and
On Wed, 2 Dec 2009 08:45:27 -0500, Frank Carmickle
wrote:
> On Wed, Dec 02, Fran??ois Legal wrote:
>>
>>
>> Hello,
>>
>> I created an extension in my dialplan so that when an incoming
>> call arrives, it rings a group of lines and then fallback to the
>> voicemail
>> if no line is answered.
>
What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions.
JM
On Tue, Dec 1, 2009 at 3:31 PM, Michael Collins wrote:
>
>
> On Sun, Nov 29, 2009 at 10:06 AM, Puskás Zsolt wrote:
>
>> Hi Guys!
>>
>> I'm using the latest svn (15711) with the default xml config. Only
>> modified
>>
On Wed, Dec 02, Fran??ois Legal wrote:
Snip...
> > voicemail config. Look at autoload_configs/voicemail.conf.xml
> >
> >
> >
> > HTH
> > --FC
> >
>
> I tried to remove the bind_meta_app from the dialplan, call the extension
> then press * when the greeting message starts, but it did not bring
As always, you are correct.
The scenario now is:
- If the caller places the callee on hold, the callee will get hold music
- If the callee places the caller on hold, the caller will not get hold music
I've uploaded a fresh pastebin here:
http://pastebin.freeswitch.org/11356
On Fri, Nov 20, 200
I have uploaded the dialplan and JavaScript files used to process calls to
MODENDP-272. I have even done a make current to revision 15755, and the blind
transfer is still failing.
_
Windows
I did check (and modify as my voicemail extension name is not vmain) the
voicemail.conf.xml, and vmain-key is *.
What I mean by remote fetching of voicemail, is that someone may dial in,
either from inside (via FXS or even SIP) or outside (via FXO), then when
reaching the voice mail to leave a mes
On Wed, Dec 02, Fran??ois Legal wrote:
> I did check (and modify as my voicemail extension name is not vmain) the
> voicemail.conf.xml, and vmain-key is *.
>
> What I mean by remote fetching of voicemail, is that someone may dial in,
> either from inside (via FXS or even SIP) or outside (via FXO),
1 - config
2 - I've done this with programming
3 - suppose programming would be needed
Here is a bunch of code, search there ''barge"
Artem
On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev wrote:
> Hi all,
>
>
>
> I’m evaluating FS for our organization.
>
> I must fulfill the followin
Frank Carmickle wrote:
> On Wed, Dec 02, Otis wrote:
>
>> Hello
>>
>> I am experimenting with FS and would like to know how to connect two
>> independent servers with user on one beinng able to call users on the
>> other. Do I set each server to be the gateway of the corresponding one ?
>>
2009/12/2 João Mesquita
> What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions.
>
> JM
>
> Thanks for catching my typo! :)
-MC
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailm
No, my voicemail extension (I have 2 actually) is called
vmain_unregistered_user, so in voicemail.conf.xml I have :
But still (and I don't even know if I'm using it the right way), I would
expect that when the voicemail greeting starts, I could press * on the
phone to call the vmain_unregistered
On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote:
>
> What would be the best way of making originate() run call through a dial
> plan
> (compared to directly going to a specified VOIP gateway). Would it be
> loopbacks, i.e. smth like this?
>
> /opt/freeswitch/bin/fs_cli -x "originate
>
> {ignore_early_m
On Wed, Dec 2, 2009 at 8:39 AM, John Platts wrote:
>
> I have uploaded the dialplan and JavaScript files used to process calls to
> MODENDP-272. I have even done a make current to revision 15755, and the
> blind transfer is still failing.
>
> John,
Thanks for keeping the guys in the loop. Just a
Hi Freeswitch users,
i'm new into the PBX world. I just installed FreeSwitch and made work
great, but one of my goal with the PBX system is to use it as a
dictation system. We were using Callweaver, and there's a Dictation
module for CW and one for Asterisk, but i can't find one for FreeSwitch
so
On Wed, Dec 02, Otis wrote:
Snip...
> Thanks.
>
> I would like all extensions on say server A to be contactable by those
> on server B and vice versa.
The example I gave you should get you started. Let us know how you get along.
Have a read through the wiki pages like
http://wiki.freeswit
1. Call recording: All (or selected) calls to the
secretary must be recorded.
Just requires an addition to the
dialplan.http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record
2. Call barge in: Assume that two subscribers are
talking to each other. Secretary makes “emergency” (for example, a
On Wed, Dec 2, 2009 at 9:21 AM, Artem Shiyanov wrote:
> 1 - config
> 2 - I've done this with programming
> 3 - suppose programming would be needed
>
>
Just to clarify, when you say "programming" there are different levels of
involvement. For example, you can do programming in C which is pretty in
On Wed, Dec 02, Fran??ois Legal wrote:
> No, my voicemail extension (I have 2 actually) is called
> vmain_unregistered_user, so in voicemail.conf.xml I have :
>
>
>
> But still (and I don't even know if I'm using it the right way), I would
> expect that when the voicemail greeting starts, I coul
On Wed, Dec 2, 2009 at 9:43 AM, François Legal wrote:
> No, my voicemail extension (I have 2 actually) is called
> vmain_unregistered_user, so in voicemail.conf.xml I have :
>
>
>
> But still (and I don't even know if I'm using it the right way), I would
> expect that when the voicemail greeting
I decided to just change the code so its more elegant to handle recursive
broadcasting so you can try again and see if that helps.
On Wed, Dec 2, 2009 at 10:35 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:
> As always, you are correct.
>
> The scenario now is:
>
> - If the call
On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle wrote:
> On Wed, Dec 02, Otis wrote:
> Snip...
>
> > Thanks.
> >
> > I would like all extensions on say server A to be contactable by those
> > on server B and vice versa.
>
> The example I gave you should get you started. Let us know how you get
On Wed, Dec 02, Fran??ois Legal wrote:
> No, my voicemail extension (I have 2 actually) is called
> vmain_unregistered_user, so in voicemail.conf.xml I have :
Also, is there a functional requirement for two different extensions to call
vmain?
--FC
___
Yes, I'm familiar with that application, check the src code for the author
=p
There has not been much of a demand for such an application but it's of
course entirely possible to develop one.
On Wed, Dec 2, 2009 at 9:00 AM, David Laperle wrote:
> Hi Freeswitch users,
>
> i'm new into the PBX wo
I need a way to start a call from the PHP script to the originating number,
tell the party on that number to hold on, start another call to destination
number, and bridge everything together. On both legs I need to pass custom
caller ID. I can of course open direct connections to VOIP gateways rig
bind to the transfer app so that it transfers the call to the vm extension
that way the current application is always interrupted and replaced.
The special "inline" dialplan lets you transfer calls right to an
application
use "inline" as the dp name and voicemail: as the extension
On Wed, Dec 2
This seems like an interesting niche project. I think that if you have
programming skills then the community can provide the PBX/VoIP knowledge to
help you get over the hump. I would recommend that you write up a document
describing all the features that this module would need to provide. Reply to
idle is a 4 letter word to a realtime application.
The core keeps a single high-priority thread to keep 1ms timing and expands
that broadcasting
to hundreds or thousand of threads who need accurate timing.
Your choppy audio is caused by linksys lying about the packet len that it's
using and we se
Hi, I got a weird issue when I dialed an extension and listen to a recorded
voice mail greeting message.
After playing a couple of time of the greeting, the FS printed the warning
of "sample rate not matching", then
send the audio to a different remote RTP port. See the log below,
2009-12-02 12:4
I'm back again with the same issue.
Now it is became worse: it reproduces occasionally.
[FS version is 1.04, test_load = 2 active calls]
I've got 2 logs: successful and not.
Here is a bad_case:
2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
java(/usr/local/freeswitch/scrip
Did you also update your wanpipe drivers and rebuild openzap again after you
upgraded it?
On Wed, Dec 2, 2009 at 2:12 AM, François Legal wrote:
> So I did some tests and still I can not see CLIP on a phone connected to an
> FXS port. Whether the call is bridged from SIP UA or from an incoming c
you should be working on SVN trunk if you are doing development, we are so
far forward from 1.0.4 we can't do debugging very easily.
I don't know all of the details of what you are trying to do but you are
hitting some race conditions because of the async nature of the socket
connection and the wa
2009. december 2. 18.35.06 Michael Collins dátummal ezt írta:
> 2009/12/2 João Mesquita
>
> > What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid
> > confusions.
> >
> > JM
> >
> > Thanks for catching my typo! :)
>
> -MC
>
Thanks for the hint I loaded mod_xml_cdr and now understand
you must only have 8k sounds so the resample is when it's playing files
try make hd-sounds-install to install 16k sounds too
On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote:
> Hi, I got a weird issue when I dialed an extension and listen to a recorded
> voice mail greeting message.
> Aft
> I love mod_xml_cdr :)
>
>
My sentiments as well.
-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
Hi, Anthony,
Thanks for your reply.
When I type the command below, I got the error,
Unknown target hd-sound-install
make[1]: *** [hd-sound-install] Error 1
make: *** [hd-sound-install] Error 2
I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail,
there are directories, 8000
Tony,
Thanks for that but now it appears that the call just gets hung up
on when the caller takes the callee off hold. Debug here:
http://pastebin.freeswitch.org/11359
Thanks again!
On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale
wrote:
> I decided to just change the code so its more el
that was make hd-sounds-install sorrry
you should also update to SVN trunk because based on the line number in your
log
its clear you are using a much older version of FS
On Wed, Dec 2, 2009 at 2:08 PM, Erwin Davis wrote:
> Hi, Anthony,
>
> Thanks for your reply.
>
> When I type the command be
Hi list;
I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and
10 years in VoIP (Cisco). I need a reference guide to start working
with
FreeSWITCH. I download the official documentation, it would need some
other configuration examples and dialplan sip device for calls.
Martin Rodr
On Wed, Dec 2, 2009 at 10:04 AM, Martin Rodriguez wrote:
> Hi list;
>
> I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and
> 10 years in VoIP (Cisco). I need a reference guide to start working
> with
> FreeSWITCH. I download the official documentation, it would need some
> other
On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote:
> Hi, Anthony,
>
> Thanks for your reply.
>
> When I type the command below, I got the error,
> Unknown target hd-sound-install
> make[1]: *** [hd-sound-install] Error 1
> make: *** [hd-sound-install] Error 2
>
> I found out that under /usr/loca
Hi, Anthony and Mike,
Thanks for your reply. The problem still exists even after I ran "make
hd-sounds install".
I will try the latest version from the SVN to see if the problem will go
away. I will let you know.
Thanks folks,
Regards,
On 12/2/09, Michael Collins wrote:
>
>
>
> On Wed, Dec 2, 2
I tried to use eavesdrop today and it did not work. The error message in the
log is:
[ERR] mod_dptools.c:334 Usage: [all | ]
I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
incorrect?
http://pastebin.freeswitch.org/11363
Thanks Lars
_
Thanks. Will let you know
Frank Carmickle wrote:
> On Wed, Dec 02, Otis wrote:
> Snip...
>
>
>> Thanks.
>>
>> I would like all extensions on say server A to be contactable by those
>> on server B and vice versa.
>>
>
> The example I gave you should get you started. Let us know how y
Sorry, svn 15753
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars Zeb
Sent: Wednesday, December 02, 2009 2:08 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Eavesdrop er
I am not sure what you are sending over the socket but you have a queued
hangup being processed on line 640 of your pastebin
are you executing any commands with a ! character in it by any chance or
executing the hangup app on purpose?
On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner <
kristia
it probably just means the uuid was not retrieved from the db when you
called the eavesdrop exten which does the lookup on the uuid for the hash
key based on what ext you hit to retrieve the most recent uuid that called
that ext.
On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote:
> Sorry, svn 157
Can I reduce resolution of that timer thread 10 times? I mean, I glanced
through the code, and see that among others (are there others?) RTP and IVR
set up their timers that are subsequently managed by this thread. RTP timers
should be eliminated by that setting you've suggested. IVR timers are se
I have read some of the archived emails about HA, loadbalancing,
failover etc and I am still a bit confused about how I could set up
some sort of resiliency with freeswitch.
My situation is much less complex than the scenarios people were
talking about and I hoping the solution is similarly much l
Is this reasonable given it was the only call in FreeSwitch at the time? How
can this situation be corrected in the future?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 02, 2009 3
Oh, looks like the timers are also used for streaming local data in
read_stream_thread(). Due to this there is always one timer active with 20ms
interval.
But wait a sec, why is freeswitch periodically trying to stream
/opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav somewhere?
Eve
I am new to FS having ditched Asterisk a few weeks ago. I have iptel
registered but I cannot get Inphonex to work. I am using the settings from
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no
avail.
The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR]
OK, I'm slow. It's music-on-hold, and it's playing non-stop like that timer
thread. Even when there are no calls. Why?
eaf wrote:
>
> Oh, looks like the timers are also used for streaming local data in
> read_stream_thread(). Due to this there is always one timer active with
> 20ms interval.
>
This is keeping track of a place in the music on hold so your hold
music does not start back up at the same place every time. If you
don't want to do this it is a module that you don't need to load and
you can get your moh from any soundfile at your choice in configuration.
Mike
On Dec 2,
In short. No, you can not for many reasons. The milisecond tic is
used throughout the code even when there is not any calls up. You can
grep for switch_cond_next if you would like to see where but it is
required to keep our global timestamp and for pacing the scheduler
among other service
As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it
could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" overrides
that.
Yeah, there is a global timestamp... It's easy to workaround that for RTP
who calls switch_micro_time_now()... But if somebody accesses
runtim
Hello,
I have Polycom phones which send only RFC-2833 (or inband which I dislike)
and they should go out to the PSTN via a Cisco gateway. The Cisco gateway
has some bug and accepts only INFO.
I did a few tests:
- Some of the phones are on different profile than the Cisco. On their
profil
Hi Guys,
I got a compilation error of skypiax_protocol.c with the latest version
r15764.
Compiling skypiax_protocol.c...
*cc1: warnings being treated as errors*
skypiax_protocol.c: In function ‘X11_errors_handler’:
skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and
code
Consider it fixed.
Committed revision 15765.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote:
Hi Guys,
I got a compilation error of skypiax_protocol.c with the latest
version r15
Hi Mathieu, thanks for the promptly reply. The error has been fixed.
However, I encounter another one.
gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99
-ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes
-Wmissing-prototypes -fvisibility=hidden -DSWITCH_API
Hi,
That one is on your side. If you changed/updated system libs it might
be worth doing another ./configure
Cheers,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote:
Hi Mathieu,
Not sure whether this error is due to the lack of libjpeg. I just double
checked, this library had been installed.
Package libjpeg-6b-37.i386 already installed and latest version
On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang wrote:
> Hi Mathieu, thanks for the promptly reply. The error has been
BTW, I forgot to update: I changed the bridge parameters to use
sofia_contact() and it solved the problem. I also fixed the presence problem
I had before with sofia_contact() (added presence_id to the bridge command).
Regards, __Yehavi:
2009/11/24 Yehavi Bourvine
> Hel
No, I didn't change or update the system libs. I just wanted to double check
whether my system has this libjpeg library. ./configure was definitely
executed before the source codes were rebuilt.
Regards,
-Jingwei
On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote:
> Hi,
>
> That one is on your
72 matches
Mail list logo
