Maybe you should give irqbalance a try:
https://irqbalance.org/
Maybe you also can assign irq 30 to a specific cpu (core):
https://cs.uwaterloo.ca/~brecht/servers/apic/SMP-affinity.txt
Am 06.11.2012 04:04, schrieb Edwin Lam:
On 11/5/12 11:59 AM, Vincent Swart wrote:
You're HDLC error is
Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Hola,
Today I started to experiment with Google Voice and Asterisk-11.0.1.
Awesome!
Following the instructions on the wiki
(https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was
able to make /
Hi
I want to know actual DAHDI channel number (pseudo), which received the
call or dialed the call. Where as when Asterisk receives a call on DAHDI
channel, it shows channel as DAHDI/i5/112-15
Is there any way / configuration to change this behavior and get actual
channel number? Earlier we
On Mon, Nov 5, 2012 at 10:43 PM, Vladimir Mikhelson v...@mikhelson.comwrote:
My practical experience shows otherwise. I am able to receive faxes on
SIP lines pretty reliably with no T.38 support. The biggest issue for
me is CED tones detection. If CED is detected then fax reception goes
I got it working so thought I'd follow up!
Here is what I had to do...
I'm using Actuate 11 and had res_fax support already
I had to download, compile, and install res_fax_digium
I had to order a Free Fax License from digium for 1 channel
I had to change faxdetect=yes to faxdetect=cng in
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 06/11/2012 02:16, Joshua Colp a écrit :
You've found a bug! I've fixed it now, though. It'll go out in the next
Asterisk 11 release or you can check out Asterisk 11 from subversion to
get it.
I have applied the patch, it now works as I
Roy,
Congratulations!
One comment and one question.
1. I would consider increasing Wait from 2 to 3 or maybe even 4 seconds
for more reliable CED detection. Some fax machines produce CED
tones not that frequently.
2. What kind of error dd you receive with faxdetect=yes in sip.conf?
Can any one suggest me what I have to do for this issue. There is no nat as
i have directly connected to internert without firewall.
Got SIP response 503 Unable to load gateways back from
xxx.xxx.xxx.xxx:5060
-- SIP/outbound-0994 is circuit-busy
== Everyone is busy/congested at this time
Darin Iv wrote:
Can any one suggest me what I have to do for this issue. There is no nat
as i have directly connected to internert without firewall.
Got SIP response 503 Unable to load gateways back from
xxx.xxx.xxx.xxx:5060
-- SIP/outbound-0994 is circuit-busy
This isn't an issue within
I want to know actual DAHDI channel number (pseudo), which received
the
call or dialed the call. Where as when Asterisk receives a call on
DAHDI
channel, it shows channel as DAHDI/i5/112-15
Is there any way / configuration to change this behavior and get
actual
channel number? Earlier we
Heya everybody.
I work on a lot of AGI/AMI/AJAM/etc.. projects and recently discovered
RELP (available via rsyslog) which is defined here:
http://www.librelp.com/relp.html
I've been pimping out (yes.. pimping) the Log dialplan application to
quickly emit a message to my local syslog which is
I have fax working but since most people and services don't know how to
Fax to Extensions,
I installed tesseract to convert the Fax to Text.
I really only need the First Page converted and will tell Faxers to make
sure they put To: Name on the cover page.
tesseract is converting the entire
I have fax working but since most people and services don't know how to
Fax to Extensions,
I installed tesseract to convert the Fax to Text.
I really only need the First Page converted and will tell Faxers to make
sure they put To: Name on the cover page.
tesseract is converting the entire
If ($_ =~ /[Tt][Oo]\:.[Nn]ame/) {
Is the way I do it.
If ($_ =~ /[Tt][Oo]..[Nn]ame/) {
Would also work
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire
Sent: Tuesday, November 06, 2012 1:51 PM
To:
On Tue, Nov 6, 2012 at 1:50 PM, Roy Abshire r...@coopvr.com wrote:
I have fax working but since most people and services don't know how to
Fax to Extensions,
I installed tesseract to convert the Fax to Text.
I really only need the First Page converted and will tell Faxers to make
sure they
Hi all,
I've been using mysql, via odbc, to store voicemails for some time now.
However, we've recently noticed that when someone leave a message of about
45 seconds or loinger, the message doesn't get stored in the database. We
do, however, receive the email notification with attachment
IIRC blobs are normally set to a limit of 65 Kb. You may need to redefine
as medium blob (16Mb) or long blob (4 Gb). 28 seconds at 44 Khz takes up
around 2.5 Mb.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent:
I am using Asterisk 11.0.0 and have a SIP user called 100. The user
registers via SIP but their network connection is cut off.
I then proceed to use AMI to send a message to them.
root@yumyum:~# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to 127.0.0.1.
Escape character is '^]'.
Asterisk
I would recommend two things. Number one would be to tweak your logger.conf
to separate out error messages. Number two would be to make your ami call do
this:
Set sip debug on
Ami action
Set sip debug off
This lets you get what you need without making a huge log file.
From:
The Asterisk Development Team has announced the release of Asterisk 1.8.18.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.18.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 10.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.10.0 resolves several issues reported by the
community and would have not been possible
Hello everybody,
A client wants to install a FreePBX infrastructure, but have all calls forward
to their cell phones rather than buying VoIP phones.
They would be doing SIP trunks over a Comcast business line. Probably maximum
6 simultaneous calls.
Any gotchas we should warn them about?
Hello listers,
I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system,
but have faced lots of problems mainly because it has lots of functions
looking for the PCI.
Have seen so many problems, I'm in fact thinking it cannot be possibly done
(at least not in a couple of weeks, by
I use Flowroute as my VOIP provider and this is exactly what I do with
all my clients. No problems except it cost me an inbound/outbound
connection each time which costs .02 = .01 inbound/.01 outbound per
minute instead of .01.
Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite
Upgrade to 11. This worked on 10.X.X
sip.conf:
register=myusername:password@nyc.teliax.net
telnet nyc.teliax.net 5060
Trying 8.14.120.23...
Connected to nyc.teliax.net.
Escape character is '^]'.
sip show registry
Hostdnsmgr Username Refresh
State
On Tue, Nov 6, 2012 at 5:33 PM, Noam Birnbaum
n...@maccentricsolutions.comwrote:
Hello everybody,
A client wants to install a FreePBX infrastructure, but have all calls
forward to their cell phones rather than buying VoIP phones.
They would be doing SIP trunks over a Comcast business line.
Can the cells be used as SIP Softphones?
Saludos/Regards
--
Ing. Gerardo Barajas Puente
Proyectos Especiales/Preventa | www.neocenter.com
T:+52 (55) 8590-9000 x 7003
On Tue, Nov 6, 2012 at 6:33 PM, Noam Birnbaum
n...@maccentricsolutions.comwrote:
Hello everybody,
A client wants to install
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 6, 2012 7:51:04 PM
Subject: [asterisk-users] 11.0.1: more sip registry woes
Upgrade to 11. This worked on 10.X.X
sip.conf:
I put ${CHANNEL(dahdi_span)} to know the span and ${CHANNEL(dahdi_channel)}
for actual channel number in incoming context of PRI.
For outbound I normally use M flag in Dial() to call a macro and check the
above variables in that macro.
--Satish Barot
On Tue, Nov 6, 2012 at 7:02 PM, Amit Patkar |
I went and checked; my database has the recording field defined as a
longblob.
Any other ideas would be most appreciated.
Mike Diehl.
Danny Nicholas da...@debsinc.com wrote:
IIRC blobs are normally set to a limit of 65 Kb. You may need to redefine
as medium blob (16Mb) or long blob (4 Gb).
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