Thanks that will be a great help
Jason White-14 wrote:
Edmar Cruz darklio...@yahoo.com wrote:
Is there a link or tutorial for the expressions format.
Anything that describes Perl regular expressions should help, and for
reference, see the pcre(3) manual page, and use the
Mike,
My latest traces that I captured were done within the FS box:
http://pastebin.freeswitch.org/11541
Thank you,
Dorn B.
From: Michael Jerris m...@jerris.com
To: freeswitch-users@lists.freeswitch.org
Sent: Thu, December 17, 2009 8:03:46 AM
Subject: Re:
I get the same result with sendmail. This used to work in 1.0.3 , and
after upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the
problem is still there.
François
On Thu, 17 Dec 2009 17:33:58 +0100,
Oliver Schönbeck wrote:
Currently it is Version 1.0.trunk (15982)
VON:
Should I open a JIRA for this?
Best regards
Peter
Peter P GMX schrieb:
Hello,
we have the following scenario:
A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For
the called FS user, call forwarding has been enabled to another PSTN
extension (B) .
Result: The calling
Hi guys (and girls)!
I'm working on a little bit of ENUM trickery and I tried doing some
(illegal) nested conditions. :-)
What I want to do is to first check enum with the ENUM application,
then depending on the answer do some stuff. Say that the domain part
of the ENUM answer is robin.nl,
I have an incomming call being answered by FreeSwitch and passed to IVR
application which rejects the call.
The call is never answered by FreeSwitch, but instead of hearing a busy
signal, the caller hears ringing.
Can anyone advise how I can get the user to hear a busy signal after call
Brian,
You haven't said what codecs are being used yet. Are the listeners
using a different codec to the speaker? If so, you're potentially
doing transcoding on every single channel, which would make CPU usage
skyrocket.
-Steve
2009/12/17 Anthony Minessale anthony.miness...@gmail.com:
What
Hi,
I am having an issue getting mod_xml_ldap to compile properly
cut-cut
making all mod_xml_cdr
making all mod_xml_ldap
Creating mod_xml_ldap.la...
/usr/bin/ld:
/home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o):
relocation R_X86_64_32S
I've got FS running on a 64 bit OS, and here is more info on the test
procedure.
I've got one server (primary) that hosts the speaker call (this is meant to
be a primary conference with a few speakers, but my test simplifies this to
just one speaker). I've got a second server (secondary) that
That depends if the call is answered and then you transfer it, you will HAVE to
set the transfer_ringback variable you can't send a 180 to the thing or a
progress and make it generate the ringback. You MUST do it yourself.
You also fail to mention if the progress is a 180 or a 183 with sdp and
Hi,
As far as I know, there are two ways to connect two freeswitch, by using ACL
or using authentication.
Also from this email history discussion, another solution is to create user
in FS B directory,then treat server B as normal gateway by adding gateway
definiton in FS A.
So my question is
I have found some ways to get presence, or rather BLF functions to work on Snom
telephones in a distributed network with several FSs. I'll post a solution on
the wiki when I have tested it further.
Anyhow, I'm using the mod_event_multicast module with the following
configuration:
On 12/18/2009 02:13 PM, Keith Laaks wrote:
Hi,
I am having an issue getting mod_xml_ldap to compile properly
cut-cut
making all mod_xml_cdr
making all mod_xml_ldap
Creating mod_xml_ldap.la...
/usr/bin/ld:
What is your dialplan on the secondary box?
On Dec 18, 2009, at 9:08 AM, Brian br...@proximosystems.com wrote:
I’ve got FS running on a 64 bit OS, and here is more info on the tes
t procedure.
I’ve got one server (primary) that hosts the speaker call (this is m
eant to be a primary
Hello Mathieu,
I assumed that apply-proxy-acl was a modifier of auth-calls, so in my
quick tests I just hard-coded the UA IP in the profile.
param name=auth-calls value=true/
param name=apply-proxy-acl value=190.218.97.83/ !-- IP of UA --
And I get:
2009-12-18 09:14:28.250929 [WARNING]
You need to add that header manually in your OpenSIPS config,
FreeSWITCH wont look in record-route/via to try to guess it.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 18-Dec-09, at 10:53 AM, Bill W wrote:
Hello
oh really,
sendmail segfaults?
if another application is crashing you need to figure that out, whatever
used to work doesnt now so you need to figure out what it was and let us
know.
On Fri, Dec 18, 2009 at 3:51 AM, François Legal de...@thom.fr.eu.orgwrote:
I get the same result with
Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like
a configuration error.
If not, I already see the title of the next Digium blog entry:
FreeSwitch scalability myth finally ends: The worst Asterisk version
ever (1.4) beating the crap of the best and latest FS.
Anyway, you
Conferencing is hardly the best place to judge performance.
Quality is a far more important goal to me in conferencing.
Lets compare who can do 48khz conferences with several 32k siren callers on
a polycom 6000, several more using G722 at 16khz and another handful of
people on g711 ulaw all at
Hello everyone!
Today's agenda is listed here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14
Also, we are going to be giving away goodies on some of the upcoming
conferences, so call in and see what we've got in store. :)
For the first 15 minutes we'll let everyone mingle and then we'll
not answering it would be the best way.
if you want to generate fake congestion you can use tone_stream:// or
gentones
On Fri, Dec 18, 2009 at 5:16 AM, bcxml bc...@hotmail.com wrote:
I have an incomming call being answered by FreeSwitch and passed to IVR
application which rejects the call.
I read through the trace, can you clarify where the missing invite is? I think
I see everything in the sofia trace.
Mike
On Dec 18, 2009, at 3:10 AM, DJB wrote:
Mike,
My latest traces that I captured were done within the FS box:
http://pastebin.freeswitch.org/11541
Thank you,
Dorn
Hi,
i have up the freeswitch with domain(eg sipserver.domain.com) name instead
of local ip, two clints are regitered with freeswitch using domain name(eg
sipserver.domain.com),
one client is making a call to other one, other clint receiving a invite
request like this
173927 3120.658532
It was of course just bad humor, I love both projects for what they are,
and I agree that both have their own advantages and inconvenients.
For example, accessing that same conference from a dahdi card could be
another goal where Asterisk would be at an advantage, as chan_dahdi is
still superior
Honestly, several years ago I accomplished this by mod'ing SER (which
became OpenSER which was then forked to OpenSIPS and Kamalio) and using
one cluster of proxies for subscriber endpoints and another for
infrastructure (so that I could keep RTP flows optimized yet support
double NAT when
yes, I understand.
My reply was to the thread in general not directed at you =p
On Fri, Dec 18, 2009 at 11:41 AM, François Delawarde
fdelawa...@wirelessmundi.com wrote:
It was of course just bad humor, I love both projects for what they are,
and I agree that both have their own advantages
Mike,
There are 2 traces in there. One is from freeswitch/sofia siptrace debug and
the other one from ngrep for your comparison.
The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion,
but it did not show in FS siptrace debug.
Thank you,
Dorn B.
On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij vi...@fx-services.com wrote:
Hi guys (and girls)!
I'm working on a little bit of ENUM trickery and I tried doing some
(illegal) nested conditions. :-)
What I want to do is to first check enum with the ENUM application,
then depending on the
Mike,
There are 2 traces in there. One is from freeswitch/sofia siptrace debug and
the other one from ngrep for your comparison.
The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion,
but it did not show in FS siptrace debug.
Thank you,
Dorn B.
Is it possible to allow/deny REGISTER requests based on the User-Agent
header? I need to know/manage what devices are registering.
Best Regards,
Jerry
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I was evaluating the technologies available, and I thought you would be
interested in my results. However, almost every other reply I get from you
to my posts, rather than being helpful, has been hostile and insulting.
My scenario is not a hypothetical one of having robots call the conference
I use a similar method (transfer to XML dialplan based on the value of
${enum_route_1}) to determine if the SIP URI is native to a particular
FS instance or if it needs to be sent off-net and it works well.
-metik
Michael Collins wrote:
On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij
Brian,
Now that you know the scale freeswotch scales to in you scenario, and
having designed a mult-server solution can you not add more server to
scale further?
As freeswitch continues to improve retest and revise your architecture
design.
Sent from my iPhone
On Dec 18, 2009, at
On 12/18/09 7:18 PM, Michael Collins wrote:
Hi Michael,
One thing you can do is create an extension that does the enum look up
and then transfers the call back into the dialplan. You could set up a
Cool idea, didn't think about that!
separate context that handles just the enum checking.
Brian, there was not one insulting word in anything I have said and as this
is a community mailing list my replies are always voiced to address the
public in general not you specifically, like I already mentioned in my last
post.
If you open a public forum on a FAQ be prepared to hear our policy.
could be possible with a code change, open a bounty on jira and someone may
do it
On Fri, Dec 18, 2009 at 12:35 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
Is it possible to allow/deny REGISTER requests based on the User-Agent
header? I need to know/manage what devices are
Try the following:
action application=hangup data=USER_BUSY/
I don't know whether it will work in your case, but here we use it to reject
a call while we want to signal that the remote party is busy.
Regards, __Yehavi:
2009/12/18 bcxml
Hi Michael,
Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure
if discussing my specific case is meant for that type of call, is it?
After Brian's suggestion to use shoutcast and local streams, I was looking
at the code for those modules. I'm not familiar with
I am more than sure there is probably plenty of room for conference
optimizations it's just a big task.
We don't have a test labbed up and an urgency to work on it. If you really
want us to pursue trying to improve the performance perhaps you can contact
us at consult...@freeswitch.org and
Hi Brian,
Have a look at this:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
- I took a quick look through the code and couldn't see any reason why
you shouldn't have a bunch of eavesdroppers listening to a single
caller. I'd be surprised if this didn't perform a lot better for
Thank you Mike for your suggestion on IRC. We did what you recommend and found
out it's the iptables issue that we thought it was not there at the beginning
since we saw the first 2 invites from the far end fine, but somehow it has
something to do with the 3rd invite.
I did close the Jira
Sounds like a plan. We will pursue it through the consult...@freeswith.org
route.
Thanks,
Brian.
From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
Sent: Friday, December 18, 2009 3:30 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users]
I think that sounds like a good idea.
It would also keep permission management simple.
-William King
On Fri, 2009-12-18 at 17:22 -0500, Frank Carmickle wrote:
Hello all
The packaging folk are interested in knowing if anyone has a problem with
having the install set up the user and group
So, it's been a while since I mentioned this project, but its finally
nearing the point where it's going to be able to go into production (and
replace my old asterisk-based platform) so I decided to dredge it up
again.
Briefly, spice telephony is a call/contact center platform that
leverages FS
I've been asked to provide some screenshots, so here's some of the
agent/supervisor interface:
http://eagle.bsd.st/~andrew/cpxshots/
Hopefully the image names are self-explanatory. In the ringing picture,
that URL pop is a configurable URL that can be used to integrate with a
CRM, in my case our
Hey Metik,
Thanks so much for your insights and your help. And yes, I was able to
append the X-AUTH-IP header with no problem. But that didn't solve the
issue. After some more research, it appears that the problem isn't with
auth-calls at all.
I disabled all auth-call directives in every
Is there any way to park a channel without causing pre-answer (resulting is
a SIP 183 Session Progress)?
Thanks,
Ron
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