Hi,
I made a silly mistake. The problem was solved. Please ignore.
Thanks.
Regards,
Pete
On Thu, Apr 17, 2008 at 12:38 PM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I followed the online installation but when I executed, make sure or
> make moh, I am getting the following errors:
>
> s
Hi,
I followed the online installation but when I executed, make sure or
make moh, I am getting the following errors:
ser:/usr/src/freeswitch-1.0.rc2/build# make moh-install
make: *** No rule to make target `moh-install'. Stop.
ser:/usr/src/freeswitch-1.0.rc2/build#
What could be the problem?
That being said, there is an open bug that the tarballs only work with
automake 1.9. That will be fixed in the next rc.
Mike
On Apr 16, 2008, at 10:54 PM, "Anthony Minessale" <[EMAIL PROTECTED]
> wrote:
rc2 and up is pre bootstrapped.
On Wed, Apr 16, 2008 at 10:46 PM, Peder @ NetworkObli
rc2 and up is pre bootstrapped.
On Wed, Apr 16, 2008 at 10:46 PM, Peder @ NetworkOblivion <
[EMAIL PROTECTED]> wrote:
> FYI, rc2 and rc3 both appear to be missing bootstrap.sh. It is in rc1.
> I copied it from rc1 to rc3 and it squawked about missing docs/COPYING
> and docs/AUTHORS, but it appe
FYI, rc2 and rc3 both appear to be missing bootstrap.sh. It is in rc1.
I copied it from rc1 to rc3 and it squawked about missing docs/COPYING
and docs/AUTHORS, but it appears to be running the rest of the setup so far.
Peder
___
Freeswitch-users ma
Hi,
Two more questions about Freeswitch. I have some application written
to send and receive fax using Hylafax and IAXModem, will I be able to
port that to Freeswitch?
What is the best way to connect Freeswitch to an E1? Does anyone has
experience with any easy-to-use, easy-to-maintain hardware
Anthony Minessale napsal(a):
> If you do the research and present us with some options I am open to
> implementing them as long as it makes sense.
>
I do what I can and lets see :-)
Thanks, Anthony!
Best regards,
kokoska.rokoska
___
Freeswitch-us
If you do the research and present us with some options I am open to
implementing them as long as it makes sense.
On Wed, Apr 16, 2008 at 1:56 PM, kokoska rokoska <[EMAIL PROTECTED]>
wrote:
>
>
> Michael Jerris napsal(a):
> > I think with the volume of calls you are handling, this is one place
>
Michael Jerris napsal(a):
> I think with the volume of calls you are handling, this is one place
> where openser will serve you better than freeswitch. You said you
> already have openser in this role, why would you not want to use it?
>
Because I need B2BUA (for topology hiding, crappy UA
Please digg and pass on
http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk
/b
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UNSUBSCRIBE:http://lists.
I think with the volume of calls you are handling, this is one place
where openser will serve you better than freeswitch. You said you
already have openser in this role, why would you not want to use it?
Mike.
On Apr 16, 2008, at 1:09 PM, kokoska rokoska <[EMAIL PROTECTED]>
wrote:
>
>
>
>
Anthony Minessale napsal(a):
> Apart from the few ways I mentioned, I don't know any other way.
Thanks, Anthony, for clarification!
> I am a
> fan of the client using STUN rather than the server compensating.
I too, but users around me not :-)
Best regards,
kokoska.rokoska
You can also elect to create a module that interfaces to a db directly but
we do not have one in tree because, as Brian said, the XML CURL was a more
flexible option for a reference implementation.
On Wed, Apr 16, 2008 at 12:11 PM, Brian West <[EMAIL PROTECTED]> wrote:
>
> On Apr 16, 2008, at 12
Apart from the few ways I mentioned, I don't know any other way. I am a fan
of the client using STUN rather than the server compensating.
On Wed, Apr 16, 2008 at 11:29 AM, kokoska rokoska <[EMAIL PROTECTED]>
wrote:
>
>
> Anthony Minessale napsal(a):
> > Yes you remove the force-contact variable
Perhaps this will help clarify.
http://www.freeswitch.org/node/117
On Wed, Apr 16, 2008 at 6:15 AM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
> This question may have come up a few times already. I am working on
> a application to provide IVR, voicemail, and tailored call routing
> services.
On Apr 16, 2008, at 12:07 PM, Pete Kay wrote:
> Hi,
> I took a look at xml_curl, is it ture that I need to write a PHP file
> to create the xml that is in the same format as the config xml files?
> Within PHP, I can go to the DB to featch the info to create the xml
> format output.
You can write
For Scalability, I'd choose Option B to limit the sip messages to increase
another x fold from a subscriber base without putting processing power into
simple overhead tasks such as options polling - responses for nat.
For Fun, both can run on the same box -- make sure your gigabit port isn't
Pete,
That's a debatable topic... Currently I'm doing over 40million minutes/month
w/out using OpenSER... Here's the no "BS" low down... OpenSER will almost
always end up faster then FreeSwitch in head to head SIP testing...
The problem comes in however with the limited feature set of OpenSER an
Hi,
I took a look at xml_curl, is it ture that I need to write a PHP file
to create the xml that is in the same format as the config xml files?
Within PHP, I can go to the DB to featch the info to create the xml
format output.
Is that correct? So, there is no direct DB access like Asterisk
real-t
Hi,
One scalability question:
Can someone provide input to the following options?
Option A: Asterisk ( for IVR, Voicemail, Conference) + Openser ( for
SIP registration )
Option B: Freeswitch + Openser
Option C: Freeswitch
So, it is definite that Asterisk is not as scalable as Freeswitch.
Among
On Apr 16, 2008, at 11:44 AM, Pete Kay wrote:
> Hi,
>
> I have studied the Freeswitch doc and can't find any info related to
> setting up the sip users, queue, ans voicemail as a realtime DB, like
> Asterisk does. Is this feature available?
>
xml_curl
> If not, would it be a bit too much work
Hi,
I have studied the Freeswitch doc and can't find any info related to
setting up the sip users, queue, ans voicemail as a realtime DB, like
Asterisk does. Is this feature available?
If not, would it be a bit too much work to write to the XML all the times?
Also, is there an Asterisk AMI-equi
Anthony Minessale napsal(a):
> Yes you remove the force-contact variable from the tag and only
> set it in the individual tag.
> That is how you can pick and choose which ones do the options.
>
Thank you very much, Anthony, for your answer!
Yes, I use force-contact per individual user, but
Pete:
Yes. All of the below, and more, are in tree and are available for your
use.
http://wiki.freeswitch.org/wiki/Mod_conference
http://wiki.freeswitch.org/wiki/Mod_fifo
http://wiki.freeswitch.org/wiki/Music_on_Hold (needs to be updated)
http://wiki.freeswitch.org/wiki/Mod_voicemail
On
Hi,
I have been seeing quite a few articles that mention Freeswitch is
much more efficient than Asterisk. What about in terms of
functionality? Can Freeswitch perform the same kind of functionality
provided by Asterisk such as Queue, Conf call, MOH, Voicemail Main,
etc?
Can anyone give some inp
It's ok, I had an idea from what you meant, but that may have been because I
was looking elsewhere and already had an idea what to expect.
I originally thought you were comparing the amount of calls per some value...
10 calls in FS for every 1 call from asterisk for the same amount of value of
http://jira.freeswitch.org/browse/MODLANG-58
On Wed, Apr 16, 2008 at 4:23 PM, Jonas Gauffin <[EMAIL PROTECTED]> wrote:
> No, I'm creating a new session:
>
> bleg = new Session();
> bleg.setCallerData("caller_id_name",
> session.caller_id_name);
>
Oh Oh! Shit... I made that sound like 10 FS boxes for 1 Asterisk box... I
have my ratios backwards... Its 1 FS box for 10 Asterisk boxes... Sorry for
the confusion this will invariably generate
K
> From: Ken Rice <[EMAIL PROTECTED]>
> Reply-To:
> Date: Wed, 16 Apr 2008 10:04:06 -0500
> To:
> S
Lets clarify here.. Its 10 Asterisk boxes to 1 FreeSWITCH box?
/b
On Apr 16, 2008, at 10:04 AM, Ken Rice wrote:
> I have done testing with FS and Asterisk in similar configurations
> on the
> same hardware, avg ratio of call handling FreeSwitch:Asterisk is 10:1
__
Hmmm,
I have done testing with FS and Asterisk in similar configurations on the
same hardware, avg ratio of call handling FreeSwitch:Asterisk is 10:1
Simple test
SIPP -> (FreeSwitch|Asterisk) Run SIPP to bring up channels doing media
playback, then call in with a SIP device to listen to the qua
http://jira.freeswitch.org/browse/MODLANG-56
On Wed, Apr 16, 2008 at 4:15 PM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> Can you please file it on jira and attach the script to reproduce it and any
> relevant logs?
>
>
>
>
> On Wed, Apr 16, 2008 at 8:17 AM, Jonas Gauffin <[EMAIL PROTECTED]>
>
Hi FreeSWecialists,
I'd like to extend title question for more details:
assuming Asterisk efficiency (concurrent 'normal' calls processing) for given
hardware
(meaning any, but fixed configuration server) as 1
[ normal call understood as no codec translation, no tricks, nothing special -
just one
No, I'm creating a new session:
bleg = new Session();
bleg.setCallerData("caller_id_name",
session.caller_id_name);
bleg.setCallerData("caller_id_num",
session.caller_id_num);
destination =
"{ignore_e
Are you using the transfer app from your javascript?
If so, and the script is running on the target channel that is being
transferred it will cause the script to exit because
transfer is effectively hangup as the act of transfer stops all execution of
apps and sends the call to the dialplan.
You c
Yes you remove the force-contact variable from the tag and only set
it in the individual tag.
That is how you can pick and choose which ones do the options.
You can also configure a force-expires on each user so the register will
reply with a very short expiry time to
trigger the client to re-re
Can you please file it on jira and attach the script to reproduce it and any
relevant logs?
On Wed, Apr 16, 2008 at 8:17 AM, Jonas Gauffin <[EMAIL PROTECTED]>
wrote:
> But why do the sound files work for incoming calls, but not for calls
> originated from a javascript?
> this feature have worked
Hello again =)
I've made a call queue in javascript where the js takes orders from a
server through a socket.
The server monitors all extensions (through eventsocket) in the queue
and tells the javascript to originate the call to a extension as soon
as it becomes idle.
Everything works fine excep
But why do the sound files work for incoming calls, but not for calls
originated from a javascript?
this feature have worked fine before (can't say which revision).
a full log is coming.
On Wed, Apr 16, 2008 at 2:59 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Well you're playing a wav file aren't
Well you're playing a wav file aren't you? That is normal the codec
for the call looks to be PCMU but to play the file it needs L16 so the
core can translate it to the caller. Plus you seem to have not posted
the full trace so I can see.
console loglevel 8
and paste that.
/b
On Apr 16,
You didn't have to do that this extension already does that! ;)
/b
On Apr 15, 2008, at 11:12 PM, sunghoon lee wrote:
>
>
>
>
>
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Hi,
This question may have come up a few times already. I am working on
a application to provide IVR, voicemail, and tailored call routing
services. The SIP registration will be handled by Openser, and
Asterisk is only doing the media function. We are talking about over
100 users.
Is Freeswit
Mike,
Thank you for answering! Actually I went into the irc yesterday
and got lots of help. It so happens that conf files were not put where
they should've been by the installer (somehow). I did a make sync (I
had the sources from two days ago) and re-did everything and its all
A-OK now! I go
ons, 16.04.2008 kl. 11.35 +0300, skrev Juha Heinanen:
> Mikael A. Bjerkeland writes:
>
> > All SIP traffic goes to the OpenSER proxy which in turn routes the call
> > with the LCR module to a FreeSWITCH server. This has proven to work fine
> > in my limited testing environment. I am using ENUM
Hi,
I am trying to look up information on the use of FreeSWITCH together
with OpenSER. I am doing a configuration on my own without much
experience on the configuration of OpenSER. In my setup OpensSER acts as
a registrar and proxy. I have multiple FreeSWITCH servers for media,
routing and PSTN te
Jonathan Palley napsal(a):
> I get the feeling that some of these edge optimizations are,
> rightfully, not the top priority for the core team.
I think so. And this why I think about to do it myself :-)
If someone other see it useful...
> If you make
> these changes I'm happy to help test a
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