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Hi,
ups, main message of me wasn't in my mail. Of course you can have it if
someone wants it.
regards
Helmut
On 24.09.2009 19:22, Michael Collins wrote:
Excellent, thanks!
-MC
On Thu, Sep 24, 2009 at 4:50 AM, Helmut Kuper
Dear Sir,
If I disable the async mode in socket, the playAndGetDigits doesn't
exit after getting the DTMF value. It exit after time out seconds. But I
need to exit when DTMF digit is got.
My subroutine call is,
$conn-playAndGetDigits(1,1,1,8000,'#',$play_list,ivr/ivr-please.wav,res,\\d+);
I beg to differ on that one.. what distro are you on? I use this all
over the place and it works so i'm concerned about this.
/b
On Sep 25, 2009, at 1:55 AM, velusamy velu wrote:
Dear Sir,
If I disable the async mode in socket, the playAndGetDigits
doesn't exit after getting the
Dear Sir,
Please pardon me for this question. Now I corrected my mistake. It is
working fine.
Thank you vrey much for your valuable help..
On Fri, Sep 25, 2009 at 12:30 PM, Brian West br...@freeswitch.org wrote:
I beg to differ on that one.. what distro are you on? I use this all over
On Fri, Sep 25, 2009 at 9:52 AM, Jason White ja...@jasonjgw.net wrote:
[Just catching up on this thread.]
William King quentus...@gmail.com wrote:
I would be more than happy to share the code I use.
Here is the git repo:
http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
Just tried that, but that simply results in *nothing* happening. It
processes the meta digit, but then it just goes on, without
transfering at all, nor if the original B-leg hangs up - that just
results in the A-leg hanging up as well.
2009/9/25 Anthony Minessale anthony.miness...@gmail.com:
in
Anthony, I am embarassed to say that it's already documented in the mod_unimrcp
wiki page now I come to look at it ... I can't understand how we didn't see
that. Must try harder.
I have however added the mod_unimrcp page to the Modules page in the wiki - it
wasn't there before, and I have
Hi,
very happy with freeswitch as a PBX/softswitch/SBC system its working solidly
for a few weeks now - just great
I have a question regarding ringback tones - custom or regular - I cant get
freeswitch to send ringback using G729
I used the following settings ( it will just play one of
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that version it
works as a charm.
for some reason i cannot get it right in 1.0.4 and trunk.
Actually, what i'm doing is to subscribe to events (within a custom module)
and try
For good measure, this is with tr...@14973
2009/9/25 Harry Vangberg ha...@vangberg.name:
Just tried that, but that simply results in *nothing* happening. It
processes the meta digit, but then it just goes on, without
transfering at all, nor if the original B-leg hangs up - that just
results
Dmitry Bely ha scritto:
It would be great if William's changes are committed to FreeSWITCH svn
repository. Anyway, debian folder in /trunk is outdated (has not been
updated since FreeSWITCH 1.0.3).
Just updated to build and package some new modules
MAx
nothing I can think of, set up a test box that is not in production
and lets figure out what is wrong.
Mike
On Sep 25, 2009, at 7:22 AM, Tihomir Culjaga wrote:
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that
should i move this to the DEV mailing list ?
T.
On Fri, Sep 25, 2009 at 4:12 PM, Michael Jerris m...@jerris.com wrote:
nothing I can think of, set up a test box that is not in production and
lets figure out what is wrong.
Mike
On Sep 25, 2009, at 7:22 AM, Tihomir Culjaga wrote:
Hi
On Fri, Sep 25, 2009 at 3:38 PM, Massimo CtRiX Cetra
ctrix...@navynet.it wrote:
Dmitry Bely ha scritto:
It would be great if William's changes are committed to FreeSWITCH svn
repository. Anyway, debian folder in /trunk is outdated (has not been
updated since FreeSWITCH 1.0.3).
Just updated
Hi guys,
I've got a strange situation that I'm at a loss to explain. With all
callers, I go through a dialplan where I check to see if they should
be a moderator, then transfer them to another which puts them into a
conference accordingly. This worked great on one server, but when I
copied the
Come on in!
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org or
via the good old PSTN at +1-213-799-1400
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FreeSWITCH-users@lists.freeswitch.org
as always, you can call skype the skypeuser skypiax5, then press 1
On Fri, Sep 25, 2009 at 6:15 PM, Michael Collins m...@freeswitch.org wrote:
Come on in!
sip:8...@conference.freeswitch.org or via the good old PSTN at
+1-213-799-1400
___
On Thu, Sep 24, 2009 at 12:29 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
moral of the story is, it's unwise to bind multiple ip to a server interface
that uses UDP signalling and the SIP spec requires
a UA to have one specific URL
Yeah, I see that now. I wasn't thinking about it
William King wrote:
It has been removed from the dependencies. Thanks go to the reporter for
finding the extra depends.
A new round of builds just went out and built. Let me know if you find
something else. Also mod_skypiax should be available.
-William King
Michael Jerris wrote:
I
Which launchpad address are you using?
I'm uploading to the freeswitch nightlies right now. And will post to
the freeswitch tagged release ppa when 1.05 comes out, or when the pre*
tags come out.
-William King
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William King wrote:
Which launchpad address are you using?
I'm uploading to the freeswitch nightlies right now. And will post to
the freeswitch tagged release ppa when 1.05 comes out, or when the pre*
tags come out.
-William King
Ah, perhaps that's why the keyserver update isn't working
This is for everyone using the ubuntu packages.
DO NOT UPGRADE PAST HARDY FOR PRODUCTION.
There is a major issue with all the newer systems. It has something to
do with the threads. In my tests I have seen a box sit at 20% cpu usage
with no calls on it at all.
This bug isn't in hardy.
I will
fixed in latest trunk,
please test
thank you
On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog d_ho...@ymail.com wrote:
Hi,
very happy with freeswitch as a PBX/softswitch/SBC system its working
solidly for a few weeks now - just great
I have a question regarding ringback tones - custom or
Hi guys,
I have solve my problem by adding
action application=ring_ready /
before
action application=set data=ringback=${us-ring}/
I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example
and noticed this line. Tried it and it works like a charm.
Thanks everybody, especially Brian
Doesn't make sense because ring_ready sends only a 180 set
ringback then pre_answer would make use do 183.
/b
On Sep 25, 2009, at 1:27 PM, Svetik VOIP wrote:
Hi guys,
I have solve my problem by adding
action application=ring_ready /
before
action application=set
Sorry for possible duplication, I am trying to figure how to reply to
the list properly.
Hi guys,
I have solve my problem by adding
action application=ring_ready /
before
action application=set data=ringback=${us-ring}/
I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example
and
Has the keyserver.ubuntu.com been uploaded with the key for the builds?
I keep getting a timeout trying to download the key to our ring.
apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 451AE93C
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will start testing soon
thank you very much
Ori
From: Anthony Minessale anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, September 25, 2009 7:30:34 PM
Subject: Re: [Freeswitch-users] Ringback when running G729 codec
keyserver.ubuntu.com is down. I have not seen any official word about it
yet.
-William King
Mark Sobkow wrote:
Has the keyserver.ubuntu.com been uploaded with the key for the builds?
I keep getting a timeout trying to download the key to our ring.
apt-key adv --keyserver
If anybody else cares, this was fixed by 14983
(http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=14983)
Thanks to Anthony.
2009/9/25 Harry Vangberg ha...@vangberg.name:
For good measure, this is with tr...@14973
2009/9/25 Harry Vangberg ha...@vangberg.name:
Just tried that, but that
There is a new function I checked in a little bit ago that lets you create any
of the SWIGTYPE_p_xxx types - all you need is a pointer to the memory to
represent whatever it is in native land. So with that, it's actually possible
to call most or all of the functions. (Yes DRK, you can now go do
does it mean, if i encode my voice files in g729 i can use mod_nativefile to
playback to a call using 729 codec?
T.
On Fri, Sep 25, 2009 at 8:30 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
fixed in latest trunk,
please test
thank you
On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog
yes
/b
On Sep 25, 2009, at 3:22 PM, Tihomir Culjaga wrote:
does it mean, if i encode my voice files in g729 i can use
mod_nativefile to playback to a call using 729 codec?
T.
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I had posted that I was interested in writing some documentation for FS or
improving upon what I've found.
I'd come up with a for dummies guide to get my crew started on FS and drew
from various sources. Probably not worth much for engineers who are working
on FS module code, but it might help
http://article.gmane.org/gmane.comp.telephony.freeswitch.user/16841
=)
Michael Collins is pushing the documentation efforts. I need to get
writing some more too.
-- William
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Hello all, I was wondering if anyone has used mod_dingaling for messaging rather than voice/video. Specifically, I would like to have FS send an XMPP message to an ActiveMQ server when it records a voicemail. Additionally I would like like to have CDR entries posted into the ActiveMQ server as
Thanks much, William.
I'll post Michael off-line and see what's what.
Regards,
Mike G.
On Fri, Sep 25, 2009 at 4:12 PM, William Suffill
william.suff...@gmail.comwrote:
http://article.gmane.org/gmane.comp.telephony.freeswitch.user/16841
=)
Michael Collins is pushing the documentation
Dear All,
How to config freeswitch for support this case ?
1. FS register to provider about 50 user account. (Each account
can't support multiple call in same time)
2. FS Check account not inuse before call out.
3. User account should be round-robin
BG
Dome C.
see chat_send api command and api_hangup_hook. In combination that
might work.
Mike
On Sep 25, 2009, at 6:07 PM, Pete Mueller wrote:
Hello all,
I was wondering if anyone has used mod_dingaling for messaging
rather than voice/video. Specifically, I would like to have FS send
an XMPP
On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote:
Dear All,
How to config freeswitch for support this case ?
1. FS register to provider about 50 user account. (Each account
can't support multiple call in same time)
Sofia gateways
2. FS Check account not inuse before
2009/9/26 Michael Jerris m...@jerris.com:
On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote:
Dear All,
How to config freeswitch for support this case ?
1. FS register to provider about 50 user account. (Each account
can't support multiple call in same time)
Sofia gateways
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