what did you have to change, to get this working ?
Jay
On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S 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]. Also I am c
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 at
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 ?
--
Sincerely
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 the
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 envir
Haha classic !!!
Can't wait for the next installment in the series !!
J
On 06/10/2009, at 1:02, Anthony Minessale
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 happens to look a lot like
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
wrote:
> Hi guys!
>
> I've tested
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" wrote:
> Dear all,
>
> In the case of asterisk PBX. I can get all the informa
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 wrote:
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used
regardless if i started freeswitch with nosql option (freeswitch -
nosql)..
I'd also seed such a torrent.
Please send the link :)
On 16/08/2009, at 6:34, João Mesquita wrote:
> I am interested and would also seed to the community
>
> On 8/15/09, Gabriel Gunderson wrote:
>> On Sat, Aug 15, 2009 at 12:13 PM, Peder
>> wrote:
>>> If you want the torrents, email me off
gt; USER_BUSY,
> NO_ANSWER,
> TIMEOUT,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 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" ;tag=ByFF2244HHvmj
To: 61xx...@yyy
connect to mysql, 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 wrote:
>
> sounds like the simplest way would be to use a web application
> ( PHP or something similar )
&
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 reg
howdy all.. ive added some extra $$$ to this existing ( seemingly
closed ) bounty
http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support
http://jira.freeswitch.org/browse/BOUNTY-4
it would be sweet if
a) a Jira admin would re-open the bo
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 pr
try looking here ...
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#PennyTel
also maybe dont use
On Sat, Apr 18, 2009 at 6:26 PM, David Robinson wrote:
> I have two gateways setup like this:
> http://pastebin.com/m41092fa6
>
> I have two dialplans setu
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 wrote:
> Hi,
>
> I’ve been seeing an issue where FreeSWITCH’s CPU and memory utilization
> climb over time; a restart of FS clears u
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 wrote:
> I wasn't going to say anything, but since somebody already mentioned
> ubuntu, I'll add that I'm using Hardy Heron LTS as well. I hav
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 inte
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.
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 wrote:
> There is also another side to make mimd
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. wrote:
> Well, I tried several
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 wrote:
> Hi Saeed,
>
> Here is a first draft of what I did to install FS on my server.
> Configuration are n
for topology hiding, use proxy media.
it means FS ignores the RTP stream totally, and just passes it through.
On Mon, Feb 2, 2009 at 5:36 PM, rod wrote:
> Hi Ken,
>
> 1) I'd like to use FS to hide topology, so bypass media is not possible
> 2) done
> 3) done
> 4) not used
> 5) i'm using this ins
use a Dynamic link...
On Wed, Dec 24, 2008 at 6:04 PM, Woody Dickson wrote:
> Hi,
>
> Is it possible to change the directory where freeswitch looks for .lua
> scripts?
> I would like to place the lua scripts in the shared drive so multiple
> freeswitch can refer to it.
>
> Thanks,
> Woody
>
> ___
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 chan
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 to define a separate log file per SIP Profile ??
Im trying to emulate the apache function of ( optionally ) having separate
logs per virtual server.
Sincerely
Jay Binks
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Freeswitch-users
can anybody suggest how to fix this ..
FreeSWITCH>show 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 22:24:20,1223295860,switch_ivr_multi_thread
>
> 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 tha
interesting discussion, ive always wondered about the use of the "Open
Source" implementation.
however I agree it is fraught with pitfalls.
the real interesting one would be the ability to use the TC400B
http://wiki.freeswitch.org/wiki/Bounty
if I could buy that card ( about $1450 ) and run 120 c
interesting to see how this compares to the likes of Lua etc..
--
Sincerely
Jay Binks
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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 of
, and
deny based upon that.
also you could run the web service on the same machine as FS, and connect on
127.0.0.1
I would imagine implementing SSL for mod_xml_curl is probably not HARD to do
, but I would bet that you will
have saleability issues if you tried to use it on a busy box.
Jay Binks
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 :)
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