:
#file 983006584-20100426-142120.alaw
983006584-20100426-142120.alaw: RIFF (little-endian) data, WAVE audio,
ITU G.711 A-law, mono 8000 Hz
-
but after i changed the active codec to the others, the output is
recognize as DATA again.
Does the .alaw-output internal codec (or whatever
Thank you Zhang Shukun,
I was wondering if it is possible to make one or ring and then stop the
call. But i don't find a way for that.
So i am doing it like .. make a call on accept wait and then hangup.
On Tue, Apr 20, 2010 at 12:34 PM, Zhang Shukun bit...@gmail.com wrote:
Dial() will
Hello !
I want to call a line and play a sound from the callee before putting it
in connection with the caller. Is this possible?
Example:
Dial(SIP/11, m) // Ring or Music...
if(call==ANSWERED) Play(announce) // Play 'announce' to the called
// To connect caller and called ?
Best
Look at option A(x) on this page:-
A(x): Play an announcement (x.gsm) to the called party.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Dial(SIP/11,mA(soundfile))
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mickael
Perfect! Thank you!
Dan Journo a écrit :
Look at option A(x) on this page:-
*A(*/x/*)*: Play an announcement (/x/.gsm) to the called party.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Dial(SIP/11,mA(soundfile))
*From:* asterisk-users-boun...@lists.digium.com
AMI writes event Ringing..., you can catch it and (via the same AMI)
send a soft hangup request.
On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Thank you Zhang Shukun,
I was wondering if it is possible to make one or ring and then stop the
call. But i don't
Hi,
I'm banging my head on this :
chmod +x /etc/asterisk/mysendmail.sh
cat /etc/asterisk/mysendmail.sh
#!/bin/sh
logger Entering $0 with arguments $*
logger $(whoami)
exit 0
cd /usr/sbin
ln -s /etc/asterisk/mysendmail.sh sendmail
tail /etc/asterisk/voicemail.conf
...
attach=yes
...
[default]
--- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote:
Hi,
I've noticed that one of my new servers (new mobo) if
drifting slowly
backwards in time (in aprox. 24 hours, system time
drifts back 5
minutes).
I have an ntpd process which is supposed to sync with
a
Thanks Motiejus Jakstsys
Thank you for the value able info i will give it a try.
2010/4/26 Motiejus Jakštys desired@gmail.com
AMI writes event Ringing..., you can catch it and (via the same AMI)
send a soft hangup request.
On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL
On 4/26/2010 7:33 AM, Vieri wrote:
--- On Sun, 4/25/10, Gordon Hendersongordon+aster...@drogon.net wrote:
Hi,
I've noticed that one of my new servers (new mobo) if
drifting slowly
backwards in time (in aprox. 24 hours, system time
drifts back 5
minutes).
On Monday 26 April 2010 07:22:33 Olivier wrote:
My understanding is that asterisk should have passed at least 2 values to
/usr/sbin/sendmail :
- one naming email's recipient (here f...@example.com)
- one naming the attached file
So I think I should have seen something like :
Entering
Hi,
After playing around with queues a bunch on 1.4.26.2, I noticed a few things,
which the patch below addresses. It addresses:
- Callers in position 0 will hear periodic/position announcements at a
very different rate than all other callers.
-- Announcements while in position 0 could be
Hello asterisk users,
I am having quite a problem finding, in the misdn.conf file, the accountcode
variable.
In sip and dahdi the variable name is account code, is there any kind of
variable to set this property in misdn.
Thanks in advance,
Alex
--
Hello,
I'm using Thomson/Technicolor ST2030S hardphones with Asterisk 1.6.
Changing from 1.6.1.18 to 1.6.2.6, I can see a change in Pickup's behaviour
and I'm a bit confused about it.
With 1.6.2.6, when extension 7791 is calling extension 7792, I can see
INVITE messages coming in and out
2010/4/26 Olivier oza_4...@yahoo.fr
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
I took a look at chan_sip.c and read this :
/* RFC3891: Replaces: header for transfer */
{ SIP_OPT_REPLACES,
Hello,
I searched this list archives and couldn't find any practical way to disable
newly introduced dialog-info based call pickups (see CHANGES file).
Suggestions ?
Regards
--
_
-- Bandwidth and Colocation Provided by
On Mon, 26 Apr 2010, Vieri wrote:
--- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote:
Hi,
I've noticed that one of my new servers (new mobo) if
drifting slowly
backwards in time (in aprox. 24 hours, system time
drifts back 5
minutes).
I have an ntpd process which is
Hi,
For IAX there is a fairly clear description of the authentication
process for inbound calls. A similar SIP document used to exist on the
voip-info wiki, but since 1.6.2 has a number of changes, I was
wondering how different (if at-all) 1.6 authentication might be in SIP
over 1.2. or 1.4
Hi Everybody,
I try to register to the Asterisk server using exosip2, this is my code :
*TRACE_INITIALIZE (6, stdout);
if (eXosip_init ()) {
printf(eXosip_init failed\n);
exit (1);
}
i = eXosip_listen_addr (IPPROTO_UDP,192.168.14.35, port, AF_INET, 0);
if (i!=0) {
Are there any other Taqua users out there?
We have a trunk to a Taqua switch through our ITSP and all outbound
calls have the ANI of the primary number on the trunk regardless of what
outbound caller-id we generate.
This is more than a little annoying, as it interferes with single-number
--- On Mon, 4/26/10, Gordon Henderson gordon+aster...@drogon.net wrote:
--- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net
wrote:
Hi,
I've noticed that one of my new servers (new
mobo) if
drifting slowly
backwards in time (in aprox. 24 hours, system
time
drifts
Been trying to get this to go but nongo :-).
I'm asking for some guidance especially if I should not be doing this on
an RT kernel.
I've installed what is supposed to be all of the requred deps.
Some factors that may be adding to my problem are:
1. this is only a test.. it's a 32bit guest OS
On Mon, 26 Apr 2010, Vieri wrote:
I ran the following and it supposedly updated my system time while ntpd
was running:
# ps ax | fgrep ntp
1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp
1623 pts/14 S+ 0:00 fgrep ntp
# ntpdate -b -u pool.ntp.org
26 Apr
Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and
1002) and a GSM Gateway with SIP extension . Two cell phones call
to the GSM Gateway number and after that they get a ring tone to dial
to the SIP extensions.
Is it possible to consider the GSM Gateway SIP extension as an
I must be missing something because this sounds REAL simple - just dial
1000, 1001 or 1002 from dialplan or do a Goto to the IVR context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
But suppose the cell phones DID number is: 11654321 and the GSM
Gateway extension has DID number:
Which is the DID number I have to use in the inbound route I create to
point to the IVR ???
Thanks again.
2010/4/26 Danny Nicholas da...@debsinc.com:
I must be missing something because this
On Mon, 26 Apr 2010, Vieri wrote:
I ran the following and it supposedly updated my system time while ntpd was
running:
# ps ax | fgrep ntp
1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp
1623 pts/14 S+ 0:00 fgrep ntp
# ntpdate -b -u pool.ntp.org
26 Apr
Hello,
I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my
FXS/FXO lines. I am running Asterisk 1.4.21.1
In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly
from SIP to FXS, but in the SIP phone I only hear a continuous noise. However,
when I
Did a little reading on this - looks like your GSM gateway is configured to
call Asterisk with second dialtone instead of direct dial to operator.
Don't know if changing that would get the DID passed through (beyond my pay
grade)
-Original Message-
From:
Hi everybody,
quite frequently I build customized RPMs with asterisk-1.4.20.1
including some special patches for it, to install the on CentOS 5.
Now I was looking to upgrade to asterisk-1.4.24.1, but the RPM-build is
not working anymore with my build environement.
In version 1.4.22 the Makefile
Probably (JIMO) had something to do with the Zaptel-to-DAHDI switch at
1.4.22.X
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorolf Godawa
Sent: Monday, April 26, 2010 4:41 PM
To: Asterisk Users Mailing
Steve Gladden wrote:
2. This is ubuntu Studio which uses an RT (realtime kernel)..
There seems to be very little aout there regarding running asterisk on
RT linux... one woudl think this would have some benefits..
Big benefits.. I've always wondered.
But moreso in a nn-virtual machine
On 27/04/10 2:21 AM, James Lamanna wrote:
Hi,
After playing around with queues a bunch on 1.4.26.2, I noticed a few things,
which the patch below addresses. It addresses:
- Callers in position 0 will hear periodic/position announcements at a
very different rate than all other callers.
Hi All,
I am using Asterisk as my pbx talking to a main proxy server.
The main Proxy server is sending unsolicited Notify Messages to the clients
after a call is established.
Is there a setting that I can tell Astersik to forward any NTY received from
Proxy to be forwarded to the End users?
Thanks for responding..
So that explains why it won't compile eh?
And wow Kevin...
I'm curious how much work would it be and would it be worth it?
I've always imagined RT kernels would be excellent for asterisk.
I've also wondered why it appears not to have been done 'out there'
Or discussed very
On Mon, Apr 26, 2010 at 11:33 AM, Olivier oza_4...@yahoo.fr wrote:
2010/4/26 Olivier oza_4...@yahoo.fr
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
I took a look at chan_sip.c and read this :
/*
On Sun, Apr 25, 2010 at 7:13 AM, russian qwerty
russian.qwe...@gmail.com wrote:
Hello, David.
Thank you for reply. But my problem is certainly in the size of JitterBuffer
of chan_local. I realy need to know how to change the size of JB (reduce).
BTW:
1. The file /etc/asterisk/dsp.conf doesn't
Hi !
Sorry if this is a long post...
I had this setup for about a year without problems :
Network A - wrv200 - internet - wrv200 - net b
The 2 networks are linked with an ipsec vpn. The 2 internet connections are
with the same cable company to minimize latency, both separates /24
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