what did you have to change, to get this working ?
Jay
On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S senaka...@gmail.com wrote:
Hi,
I got it working.
Can somebody explain me this error:
2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR:
[Missing local host].
I'm interested in what the upper limit would be, when expecting a
performance improvement with sofia profiles.
For example let's say I were to direct connect to customers ( layer
2 ) with a .1q trunk coming in to fs and a Sofia profile for each
customer. Am I going to hit a bottleneck
Guys,
im after info from people with experience with AudioCodes Mediant 2k PRI
Gateways.
specifically how well they inter-op with Freeswitch, and how compliant their
SIP stack is.
I guess the bottom line is, would you recommend these gateways or would you
suggest something else ?
--
if you suspect 15431 to have caused this, then revert to 15430 and see
if the problem exists.
if you can narrow do the bug to a specific svn revision, then you
greatly assist in the resolution of the issue.
apart from that im not much help sorry.
maybe someone else can lab it up and see if
I believe OBDC is the official way..
however id love look at doing this in a higher performance way, without the
single point of failure..
local memcache, in front of OBDC or something ??
not 100% sure of it, but just using a single central database is a little
bit of a concern in a carrier
Haha classic !!!
Can't wait for the next installment in the series !!
J
On 06/10/2009, at 1:02, Anthony Minessale
anthony.miness...@gmail.com wrote:
neat,
Here's some suggestions for your next ones. =p
Have them standing around the hologram trying to destroy the Death
Star(tm) that
A few thing stuck out to me ...
Mainly 50 calls and transcoding speex.
Try it again with g711 and see how you go.
Also not sure windows 7 is going to perform as good as other options,
could be wrong though .
Jay
On 17/09/2009, at 3:56, Роберт Тверитнер
siniy...@gmail.com wrote:
Hi
Check out INFO
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info
Throw that in your dialplan the look at your logs... You should find
what your after..
Jay
On 29/08/2009, at 16:45, Thangappan.M thangappan...@gmail.com wrote:
Dear all,
In the case of asterisk PBX. I can
Anthony can you ( or anyone else alao ). Please elaborate on what
makes centos 5.3 o much better for Freeswitch.
Is there some specific library vesiion on centos that makes a massive
difference ?
Reason I ask ... I personally only have a preference for debian,
but others may have policy
Everytime someone asks this , the resounding answer is use a 64bit os..
No question
Jay
On 25/08/2009, at 23:19, Tihomir Culjaga tculj...@gmail.com wrote:
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used
regardless if i started freeswitch with nosql option
I'd also seed such a torrent.
Please send the link :)
On 16/08/2009, at 6:34, João Mesquita jmesqu...@gmail.com wrote:
I am interested and would also seed to the community
On 8/15/09, Gabriel Gunderson g...@gundy.org wrote:
On Sat, Aug 15, 2009 at 12:13 PM, Pederpe...@networkoblivion.com
,NO_ROUTE_DESTINATION,CALL_REJECTED,USER_NOT_REGISTERED
You can know more about the hangup causes here:
http://wiki.freeswitch.org/wiki/Hangup_causes
Regards,
Raul
On Sat, 2009-07-18 at 13:49 +1000, Jay Binks wrote:
I have an upstream provider that utilizes a load balancer that spits
back 302
I have an upstream provider that utilizes a load balancer that spits
back 302 redirects with contact headers
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:
5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx
From: test
sounds like the simplest way would be to use a web application ( PHP or
something similar )
that handles the users Directory.. that way you can keep your DB
exactly the same and just pull the required fields.
Jay
On Mon, 2009-07-06 at 15:05 +0530, ram wrote:
Hi
I am using Opensips as
, however Im not
a fan of OBDC.
Jay
On Mon, 2009-07-06 at 16:33 +0530, ram wrote:
On Mon, Jul 6, 2009 at 3:53 PM, Jay Binks jaybi...@gmail.com wrote:
sounds like the simplest way would be to use a web application
( PHP or something similar )
that handles
Ive used these in the past.
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html
sound fine, work well...
reliable etc etc..
things to watch out for... :
* cant send your own caller ID from them ( in my experience its locked to
the sim )
* your
try looking here ...
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#PennyTel
also maybe dont use
param name=expire-seconds value=600/
param name=extension value=1000/
On Sat, Apr 18, 2009 at 6:26 PM, David Robinson pawzl...@gmail.com wrote:
I have two
what happens in your dialplan ?
is is possible that you execute a script on each call, thats not being
exited ?
Jay
On Tue, Mar 17, 2009 at 6:19 AM, Chris Fowler ch...@fowler.cc wrote:
Hi,
I’ve been seeing an issue where FreeSWITCH’s CPU and memory utilization
climb over time; a restart of
I personally like and use Debian ..
all my boxes are debian 4...
havnt looked at using debian 5 yet.
Jay
On Sat, Mar 7, 2009 at 8:07 AM, Stephen Crosby stevecr...@gmail.com wrote:
I wasn't going to say anything, but since somebody already mentioned
ubuntu, I'll add that I'm using Hardy Heron
Back in November, Brian ( BKW ) was raising money to get new sounds recorded
...
intending to have them for the 1.0.2 release..
I wonder if they made it in, or if they are still coming ...
Jay
On Tue, Feb 17, 2009 at 8:19 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
There is also
wow... this is awesome !
good job mate.
On Wed, Feb 18, 2009 at 9:43 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
FreeSWITCH now compiles in AsLinux:
http://www.astlinux.org
AstLinux with the new bootloader Runnix (or you could just use
syslinux) boots from flash. It
a man
geez im interested in this ..
I hope it ends up kicking ass ! :)
Congrats, you are awesome.
Jay
On Wed, Feb 18, 2009 at 11:41 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
I really need to work on that name but in the meantime it seems like
people are
another thing to try here...
is to put FS in RTP proxy and bypass mode.
http://wiki.freeswitch.org/wiki/Bypass_Media
it would be interesting to see if your still experiencing this problem in
either of those 2 modes.
Jay
On Mon, Feb 16, 2009 at 12:04 PM, Paul D. pa...@versafon.com wrote:
Rod,
that wiki article is Awesome !
real good to see guides with start to finish steps.
cant wait to see the next installment of your guide :)
Jay
On Tue, Feb 3, 2009 at 12:33 AM, rod kawa...@laposte.net wrote:
Hi Saeed,
Here is a first draft of what I did to install FS on my server.
id also love to get any info from the RTCP...
even have this in the XML CDR would be great..
would love to derive quality stats for calls based on RTCP
Jay
On Tue, Dec 9, 2008 at 2:37 PM, Jonathan Palley [EMAIL PROTECTED] wrote:
I'm curious to start a discussion on being able to query a
log messages.
Im not sure whats involved, but id throw a little money towards seeing this
added.
anyways... I guess its one to add to jirra..
Jay
On Mon, Oct 20, 2008 at 10:50 PM, Michael Jerris [EMAIL PROTECTED] wrote:
On Oct 20, 2008, at 8:42 AM, jay binks wrote:
Is there an easy way
can anybody suggest how to fix this ..
FreeSWITCHshow calls
created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid
2008-10-06
But with modern CPU's 120 channels isn't that much of a stretch is it?
Your probably right...
Not sure how easy that would be since you have to use the latest dhadi
release to interface with it.
Bummer..
I just thought.. if it were as simple as tweaking the Zaptel bindings to
make that
while I totally agree...
id love to see it... but Id also love the core devs to keep working on more
important stuff..
after FS is setup and running, your not going to need to attach to the
console to do stuff.
and when you do, fs.pl is there for the occasional usage.
so yea... I guess its one
I realize 1.0.1 is not a scheduled release, and that there is no set release
date..
however Im hanging out for it to release on my production systems ( need
some of the bug fixes )
I realize I COULD just use SVN head, but for a matter of a few days I
decided to wait..
its now been a few weeks :)
30 matches
Mail list logo
