Adding /n partly solved the problem. The two calls are getting connected,
but LCR is not working. The 2nd call goes out on the same trunk as the first
call ( 1st call was landline, 2nd was mobile, two different routes )
Tamas
2011/2/8 fai...@vopium.com
Just verified I faced the same issue
Hi
We're using asterisk 1.4.17 debian package soon moving to 1.8 rpm
package.
When using MixMonitor to do call recordings, for outbound calls I have
been using the channel variable SIPURI to get the originating SIP
extension name. I have now stumbled across a few files where the SIP
extension
Tilghman,
When you say reformat the audio, do you mean sample rate and bits per
sample, etc...or do you mean the format in which each packet of data is
structured ? I just want to make sure I know which one I'd be dealing with
if recording a call that was using one of the higher quality codecs
I have a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send
SMS messages over VoIP. My Asterisk 1.4 installation drops these
messages and returns a failure condition to the phone:
[Feb 9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received
message to
On Wed, 9 Feb 2011 00:01:49 -0600, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Nice! That was some good reading!
Unfortunately, I checked how the uClinux kernel was configured for
compiling, and the inotify is indeed selected by default :-/
Linux Kernel Configuration
File systems
ashishchauhan07...@gmail.com has chosen to send you a free movie ticket, up to
$10 value.
Sit back, relax, and enjoy! In an effort to spread the word about our great
products we are allowing our members to send free movie tickets to their
selected friends at no charge to our members. This
On Wed, 09 Feb 2011 11:47:09 +0100, Gilles codecompl...@free.fr
wrote:
Unfortunately, I checked how the uClinux kernel was configured for
compiling, and the inotify is indeed selected by default :-/
Greping the Asterisk source code for inotify only returned a couple
of hits, in binaries
--- On Tue, 2/8/11, Jonathan Thurman jonat...@thurmantech.com wrote:
It depends on your configuration. If you use Asterisk
Realtime to
store SIP registrations, then the database will contain
information on
how to contact the device (fullcontact, ipaddr, and port
fields).
Then on a
2011/2/5 Roberto Piola roberto.pi...@visiant.it
In Italy, you must enable overlapdial=yes
Is this relevant for incoming calls, as OP asked ?
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New to
I'm assuming you haven't googled for solution, please go through
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and
extra links in that article.
If that article were not helpful, please provide more information of
you setup, such as what analog card are you using, are you using
Hello,
I tried this piece of extensions on my Asterisk 1.8:
exten = 679,1,NoOp(start)
exten = 679,2,AGI(/var/lib/asterisk/bin/test.py)
exten = 679,3,NoOp(--- end ---)
exten = 679,n,Hangup
where test.py executes a queue command.
The strange thing is my CLI never shows the '--- end ---' string.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DeadAGI
read the part at the bottom about ignoring sighup, if you're using a
later version i think there is a option like agisighup that you can
use in the dialplan
--
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On Wed, 9 Feb 2011, gincantalupo wrote:
I tried this piece of extensions on my Asterisk 1.8:
exten = 679,1,NoOp(start)
exten = 679,2,AGI(/var/lib/asterisk/bin/test.py)
exten = 679,3,NoOp(--- end ---)
exten = 679,n,Hangup
where test.py executes a queue command.
The strange thing is my CLI
On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote:
Tilghman,
When you say reformat the audio, do you mean sample rate and bits per
sample, etc...or do you mean the format in which each packet of data is
structured ? I just want to make sure I know which one I'd be dealing
with
On Wednesday 09 February 2011 06:28:43 Gilles wrote:
On Wed, 09 Feb 2011 11:47:09 +0100, Gilles wrote:
Unfortunately, I checked how the uClinux kernel was configured for
compiling, and the inotify is indeed selected by default :-/
Greping the Asterisk source code for inotify only returned a
We have a customer who wants to forward an extension to their cell phone,
if and only if that extension is unavailable, or when the Dial() command
times out. However, should the Dial() command return busy it should go
to voicemail instead.
As far as I know, the dialplan doesn't support this.
Hi List,
Would someone can to explain me the main difference in SWITCHES or
ESWITCHES in AEL.
context default {
switches {
DUNDi/e164;
IAX2/box5;
};
eswitches {
IAX2/context@${CURSERVER};
};
};
All the best,
Thiago
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Wednesday, February 09, 2011 1:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Defining what an extension should do after
It's nice to know that you've tried this and are presenting me with a
proven solution.
FYI, this doesn't work. Neither do any of the following variations:
exten = 27,1,Dial(SCCP/foo,10)
exten = 27,n-BUSY,Voicemail(27)
exten = 27,n-NOANSWER,Dial(DAHDI/g1/5551234)
exten = 27,n,Hangup()
or
exten
Un-top-posting...
On Wed, 9 Feb 2011, Ernie Dunbar wrote:
We have a customer who wants to forward an extension to their cell
phone, if and only if that extension is unavailable, or when the
Dial() command times out. However, should the Dial() command return
busy it should go to voicemail
On Wed, 2011-02-09 at 14:37 -0800, Ernie Dunbar wrote:
On Wed, 9 Feb 2011, Ernie Dunbar wrote:
We have a customer who wants to forward an extension to their cell
phone, if and only if that extension is unavailable, or when the
Dial() command times out. However, should the Dial() command
On Wed, 9 Feb 2011 12:33:00 -0600, Tilghman Lesher
tilgh...@meg.abyt.es wrote:
Inotify for spoolfiles is supported starting in Asterisk 1.8.
Thanks for the tip, but I'm stuck with a 1.4 because it must be
patched to run on uClinux :-/
A possible explanation for this issue could be that Asterisk
Just recently I noticed that my Asterisk 1.8 server is giving the
following error at startup:
[Feb 9 17:48:56] WARNING[7968]: loader.c:387 load_dynamic_module: Error
loading module '��Է�Vi': /usr/lib/asterisk/modules/��Է�Vi.so: cannot
open shared object file: No such file or directory
On Wednesday 09 February 2011 13:31:36 Thiago Maluf wrote:
Hi List,
Would someone can to explain me the main difference in SWITCHES or
ESWITCHES in AEL.
context default {
switches {
DUNDi/e164;
IAX2/box5;
};
eswitches {
On 2/9/11 6:55 AM, Vieri wrote:
I'd like to do that without Realtime (or with Realtime+FreePBX) or with any
other means that doesn't require more than 2 servers (2 asterisk boxes)?
we use drbd nfs cluster to store asterisk's ASTDB voice mail
files but that would involve installing 2 extra
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of
Hi,
Under sip-out why do you have secret, fromdomain and NAT commented out ?
Also it seems like Asterisk is re-transmitting which means it seems like it
is not getting any response from your ISP. It could be a firewall issue, it
could be your ISP. If your ISP refuses to work with you you may
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, February 09, 2011 21:38
Subject: Re: [asterisk-users] Defining what an extension should do after
theDial() command
Hi,
I had the same issue as well but for some reason I was unable to reproduce.
Please have a loo at: https://issues.asterisk.org/view.php?id=18682
Regards,
Dovid
- Original Message -
From: M S
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 09, 2011 06:11
Anyone here who has configured zaptel/dahdi for a singtel E1 line?
What are the settings for coding, framing, line type and switchtype?
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New to Asterisk?
On 02/10/11 14:00, Dovid Bender wrote:
Hi,
Under sip-out why do you have secret, fromdomain and NAT commented out ?
Also it seems like Asterisk is re-transmitting which means it seems
like it is not getting any response from your ISP. It could be a
firewall issue, it could be your ISP. If
The settings you are asking varies in different countries and providers. You
need to contact you provider for it.
From: Roi Stork
Sent: Thursday, February 10, 2011 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] zaptel/dahdi settings for singtel E1
There are some flags in general settings of dialplan which enable/disable
modify this behaviors of dialplan. Have a look on sample extensions.conf for
general tab settings. I will see if I can have time today to tell you exact
parameter name.
From: Dovid Bender
Sent: Thursday, February 10,
On Wed, Feb 9, 2011 at 12:31 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:
On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote:
Tilghman,
When you say reformat the audio, do you mean sample rate and bits per
sample, etc...or do you mean the format in which each packet of data
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