Just to add data to this:
PowerTOP 1.11 (C) 2007, 2008 Intel Corporation
Collecting data for 15 seconds
Detailed C-state information is not available.
P-states (frequencies)
2.34 Ghz 0.0%
2.00 Ghz 100.0%
Wakeups-from-idle per second : 405.4interval: 15.0s
no ACPI power
Hi
I can also help with the translation into Spanish.
Spanish is my native language.
I can be contacted at:
rjpereyra (at) gmail (dot) com
roberto
--
The best dedicated servers - LiquidWeb
http://www.liquidweb.com/?RID=contenid
___
Hi,
I have been testing inbound calls to a Nokia phone with handover to a
cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and
had to set rtp-timeout to a very low 6 seconds in order to get fast
handover. This introduces an interesting side-effect that hangs up calls
even in the
Chris Danielson wrote:
Excellent thanks, this is what I was looking for.
One last question if you don't mind; is there anyway to pull the caller
out of a fifo after a certain time either from api or by setting a
variable (eg. the destination didn't answer after sometime, so carry on
in
learn to think 4th dimensionally =D
Add one member with a | sep list in the dial string.
On Mon, Apr 27, 2009 at 11:35 PM, seven dujinf...@gmail.com wrote:
Hi,
I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin
Are you geting 183+sdp from the nokia?
the media timer only operates once media is established and only
counts against you if the channel is being read from and that does not
happen until you get a 183 or 200 w/sdp
try putting a debug line in switch_rtp.c around 1520
printf(MISSED PACKETS
Thanks Brian. I'd like to make sure I finally got how dialplans, contexts,
domains, SIP profiles, and extensions in the directory work together:
Freeswitch supports different SIP profiles in conf/sip_profiles/
(internal.xml, external.xml, etc.), which are loaded through
Start with installing the windows sdk.
error C2143: syntax error : missing ']' before 'constant' C:\Program
Files\Microsoft SDKs\Windows\v6.0A\include\ras.h
http://msdn.microsoft.com/en-us/windows/bb980924.aspx
--dave
http://dave.thehorners.com/tech-talk/programming
On Tue, Apr 28,
Actually, you might have it it doesn't look like the download
scripts are workingcould be file permissions or strange data.
I'd delete the directory and do a fresh checkout build too.
--dave
http://dave.thehorners.com/tech-talk/programming
On Tue, Apr 28, 2009 at 8:52 AM, David A. Horner
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed dial string is hard to do round robin hunting,
as we don't want the first agent always busy while others have nothing
to do. It is possible to add/delete
What I want to discuss concerns mostly Sofia-SIP, but not primary FreeSWITCH.
Surely it would be better to use Sofia's list instead and I already did it, but
have got no response so far.
Overall Sofia-SIP is a good library, which works reliable enough on Linux, but
Windows port causes me some
dujinfang ha scritto:
It is possible to add/delete members using another script
I'm not such expert but i think you can use variables everywhere in FS.
So your member config instead of having a static string
us...@domain|us...@domain probably you can use a variables.
Then you script just push
keep in mind mod_fifo is not a call center app. it's a simple *fifo* queue
hence the name.
On Tue, Apr 28, 2009 at 8:30 AM, dujinfang dujinf...@gmail.com wrote:
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed
I also installed the Nokia Configuration Tool (V3).
Which settings did you apply for getting the crypto line?
So far I only got TLS to work, there is no crypto line so far.
Best regards
Peter
Ognjen Seslija schrieb:
Hi,
after installing Nokia Configuration Tool I managed to get E61i to
The scenario I was referring to was actually an outbound call from a
locally registered SIP phone to a cellphone. The same thing happens
whether I use a SIP or PRI trunk. After 6 s it hangs up.
I get SDP on 183 no matter whether I'm calling a cellphone or a fixed
line. I also get ringing
Hi,
there is a edit online option to use to change the option on the phone.
You have to first install SIP Voip Settings Tool from the Nokia forum, and
then change the option Secure call to prefered from NCT.
Ognjen
On Tue, Apr 28, 2009 at 4:32 PM, Peter P GMX prometheus...@gmx.net wrote:
I
Thank you for the detailed implementation, comments follows.
On Apr 28, 2009, at 10:24 PM, Anthony Minessale wrote:
keep in mind mod_fifo is not a call center app. it's a simple *fifo*
queue hence the name.
Yeah but FS is more than a *, it should be able to do a call center
like job.
as soon as FS sees 183 it expects media.
if they send 183 and no media it will most certainly timeout
On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland
mik...@bjerkeland.com wrote:
The scenario I was referring to was actually an outbound call from a
locally registered SIP phone
dujinfang ha scritto:
Yeah but FS is more than a *, it should be able to do a call center
like job.
I think if you open a bounty you can get some nice feedback about this.
Antonio
___
Freeswitch-users mailing list
Thanks! I'll notify them of the problem and see if there's a way around it.
2009/4/28 Anthony Minessale anthony.miness...@gmail.com
as soon as FS sees 183 it expects media.
if they send 183 and no media it will most certainly timeout
On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander
I just wanted to let everyone know that there are a few blog posts out there
that you FreeSWITCHers might find interesting. Please check out these two
stories on the main FreeSWITCH page:
Another Interesting Use For FreeSWITCH:
OpenSimhttp://www.freeswitch.org/node/175
DIDX Interview With
FYI, here's the correct link to the DIDX interview story on the main page:
http://www.freeswitch.org/node/176
Sorry for the broken link. (Thanks moy!)
-MC
-- Forwarded message --
From: Michael Collins m...@freeswitch.org
Date: Tue, Apr 28, 2009 at 9:27 AM
Subject: FreeSWITCH In
Hi All,
Could you please refer me to a best DID provider which also works
perfectly with FreeSWITCH. I would need 50 channels per DID number.
Thanks.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works
with any major telco.
- Original Message
From: Saeed Ahmed saeedahmad1...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:09:42 PM
Subject: [Freeswitch-users] DID
Please check here for more information.
http://www.voip-info.org/
- Original Message
From: Saeed Ahmed saeedahmad1...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:09:42 PM
Subject: [Freeswitch-users] DID Provider
Hi All,
Could you please refer
Hi,
But which one would you prefer? In terms of cost and quality.
Thanks.
Paul wrote:
FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH
works with any major telco.
- Original Message
From: Saeed Ahmed saeedahmad1...@gmail.com
To:
https://teliax.com/?referral_code=11
Teliax can provide 50 Channels per did.
Contact me for more information.
Thanks,
Geoff Love
303-629-8304
On Tue, Apr 28, 2009 at 11:09 AM, Saeed Ahmed saeedahmad1...@gmail.comwrote:
Hi All,
Could you please refer me to a best DID provider which also
Go with Verizon. They're reliable and available anywhere in the US.
- Original Message
From: Saeed Ahmed saeedahmad1...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:21:46 PM
Subject: Re: [Freeswitch-users] DID Provider
Hi,
But which one would
The best quality and cheapest for all applications serving all markets
around the world, all at the same time? Tall order.
It's easier for people to give meaningful answers if you give some hints
as to little details such as what country you're talking about (though
the working assumption of
Yup, I would need it for inbound call center services.
Thanks for your detailed reply.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jon
Radel
Sent: Tuesday, April 28, 2009 8:07 PM
To:
Is there a way to skip the person at extension... announcement when
forwarding a call to voice mail, so that it starts recording
immediately, or just says something like start recording...?
Also, is it possible to turn off voice mail play back feature, meaning
when somebody tries to leave a
This macro:
macro name=voicemail_play_greeting
input pattern=^(.*)$
match
action function=play-file data=voicemail/vm-person.wav/
action function=say data=$1 method=pronounced
type=name_spelled/
action function=play-file data=voicemail/vm-
The link in the post refers to pre5, not pre6.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, April 28, 2009 12:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users]
But is there a way to keep original macro as well and trigger them on
demand from a dial plan?
Brian West wrote:
This macro:
macro name=voicemail_play_greeting
Fixed.
/b
On Apr 28, 2009, at 3:38 PM, Alex Rambau wrote:
The link in the post refers to pre5, not pre6.
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
You could just record a greeting and copy a file thats 1ms long over
the top of it.
/b
On Apr 28, 2009, at 3:46 PM, paul.degt wrote:
But is there a way to keep original macro as well and trigger them on
demand from a dial plan?
Brian West
br...@freeswitch.org
-- Meet us at ClueCon!
oh, thank you Antonio. I think it would be better to collect more
ideas before open a bounty. And I more interested in playing(including
patching the code) with that than use the function.
On Apr 28, 2009, at 11:39 PM, Antonio Gallo wrote:
dujinfang ha scritto:
Yeah but FS is more than a
Hi List,
I have searched and searched and have found nothing. Id like to be able to
keep track of current calls and the CPS and then chart these via Cacti. Is
their some a SNMP plugin that I have not seen for FS that can do some of
these things?
Also, if I have to make some custom plugin that
every second is way too high of a resolution. 10 seconds 20 seconds
are more sane!
/b
On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote:
lso, if I have to make some custom plugin that montiors output the
of show channels /show calls via one of the API's, anyway we can get
a realtime
will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU?
Only reason id like to hit every second so I could get some sort of attempt
on the call setups and call transactions.
Has anyone even done anything like this before?
Thanks
On Tue, Apr 28, 2009 at 7:16 PM, Brian West
Connect to the socket and subscribe to the heartbeat. The heartbeat
event fires every 20 seconds and contains the session count.
On Tue, Apr 28, 2009 at 21:19, Ron McCarthy ronmc...@gmail.com wrote:
will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU?
Only reason id
On Apr 28, 2009, at 9:19 PM, Ron McCarthy wrote:
will hitting it every 10 or 20 seconds hurt performance / eat a lot
of CPU?
No its just silly in my opinion if you want that then hook on event
socket using ESL and write something to register callbacks for each
event and collect stats.
That's really I want that looks perfect, anyways I can get stats for a
gateway? Far as the calls-in and calls out?
If not the heartbeat looks like I could writ esomething up and use that to.
On Tue, Apr 28, 2009 at 7:33 PM, Brian West br...@freeswitch.org wrote:
On Apr 28, 2009, at 9:19 PM,
I know this isn't the place to report bugs; unfortunately, the Jira Web
interface isn't working for me due to accessibility issues. (If there is an
alternative way to submit reports that could be efficiently handled by the
developers, let me know).
A few weeks ago I reported problems with the
44 matches
Mail list logo
