I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls
On Wed, Sep 14, 2011 at 7:04 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
I have carried out the below steps
[root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1
obd-demo.alaw
sox: Output file obd-demo.alaw: using sample rate
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends
Hey,
The callee server is complaining too loud Call from '2765' to extension '*
1166:password*' rejected because *extension not found*.
Try changing the Dial string as DIAL(SIP/asterisk-callee/${EXTEN}) or w/e
extension you require in place of ${EXTEN}
Let me know what changes.
Also this is a
On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote:
Hi all,
I presume i made a silly mistake while filling a database
But while googling on the results, i came across a lot of messages about
the layout of app_data in case of goto and dial statements.
(mostly about using the old |
Hi
I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as
my resource
http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html
when I execute the following command
snmpwalk -On -v2c -c public 127.0.0.1 .1.3.6.1.4.1.22736
I get the following response
Hi all,
using asterisk 1.4 or 1.6, I face a problem with the read command.
I call my asterisk box which ask me to enter the number I wish to call.
Problem is that if I make a mistake in the number and correct it on the
phone keyboard (smartphone under android, the same with nokias series
E),
the provider/carrier changed his setting how to submit DTMF to our
asterisk. It was set to send SIP-INFO and rfc2833 to send only
rfc2833
Kristijan
2011/9/13 virendra bhati virbh...@gmail.com:
Hi ,
What was the solution of that problem ? Did provider change the setting at
there end or else ?
Stephen H. Gerstacker wrote:
I'm just a simple programmer who happens to be the only IT guy in the office.
And I'm just an IT guy, that started learning (And still am) about phone
systems around 10 years ago.
I was just going through the process of elimination, the differences
between our
Stephen H. Gerstacker wrote:
Is there a big difference between the two?
From what I've read, a DMS100 can redirect a call off of your system,
meaning that if you have an inbound call and you want to redirect it to
a different number, the DMS100 will redirect the call and take your
system
On Wed, 2011-09-14 at 09:00 +0100, Ishfaq Malik wrote:
On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote:
Hi all,
I presume i made a silly mistake while filling a database
But while googling on the results, i came across a lot of messages about
the layout of app_data in
/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001-20110914-163803.wav|bW(2)|/usr/bin/lame
/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001-20110914-163803.wav
/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001
On Wed, 2011-09-14 at 12:01 +0200, Hans Witvliet wrote:
On Wed, 2011-09-14 at 09:00 +0100, Ishfaq Malik wrote:
On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote:
Hi all,
I presume i made a silly mistake while filling a database
But while googling on the results, i came
On Wed, Sep 14, 2011 at 5:27 AM, Dale Noll dn...@wi.rr.com wrote:
On 09/13/2011 07:49 PM, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/**custom/${TOPMENU})})
im trying to see if a file is available
On Wed, Sep 14, 2011 at 4:08 AM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 14 Sep 2011, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/**
custom/${TOPMENU})})
im trying to see if a file is
Hello,
I do the following in a macro in the dialplan :
exten = s,n,MYSQL(Connect connid localhost user password AsteriskDB)
exten = s,n,MYSQL(Query resultid ${connid} UPDATE custDB SET active=1
WHERE routeID=${ARG1} AND nr=1)
exten = s,n,MYSQL(Disconnect ${connid})
But nothing changes in my
I expect that your same query when executed directly on MySQL console
executes successfully ! Normally errors in DB queries are printed on CLI but
apparently there is nothing wrong.
On Wed, Sep 14, 2011 at 5:51 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
Hello,
I do the following in
On 09/14/2011 02:51 PM, Jonas Kellens wrote:
Hello,
I do the following in a macro in the dialplan :
exten = s,n,MYSQL(Connect connid localhost user password AsteriskDB)
exten = s,n,MYSQL(Query resultid ${connid} UPDATE custDB SET active=1
WHERE routeID=${ARG1} AND nr=1)
exten =
Came in this morning to more of the same:
PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/23 got hangup, cause 81
Also, I got a lot of this as well:
[Sep 14 05:54:05] WARNING[16624]: sig_pri.c:1054 pri_find_dchan: Span 1: No
D-channels available! Using Primary channel as
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi James,
How did you resolve this issue?
[Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect:
SRTP unprotect: authentication failure
[Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect:
SRTP unprotect: authentication
Hello,
For the moment I have the following settings in my sip.conf. I want to
optimize them to archive the following things:
- for the moment all my users will re-register too often. I want that only
lagged users to re-register quickly.
- check from time to time all users but no too often to see
+1 Dale - although it would be a good idea for OP to know the in's and out's
of both System and AGI, this is a simpler way for him to catch a fish today.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dale
Since the Read command takes in its input 1 digit at a time (I don't think
this changes in 1.8 or 10.X either), your best option here would be to
follow the read with a press 1 to accept or 2 to re-enter IVR
[get-number]
Exten = s,1,Read(number,prompt1,10,skip,1,10)
Exten =
So any software echo canceller available in dahdi isn't good enough?
2011/9/13 Kevin P. Fleming kpflem...@digium.com
On 09/13/2011 08:56 AM, Gustavo Santos wrote:
I'm trying to use Asterisk as a PSTN simulator to run performance tests
for echo cancellation algorithms. I'm using the following
Stephen H. Gerstacker wrote:
Came in this morning to more of the same:
Then, if you have the ability, I'd drop 1.2 back into place and see if
it's happy. But, my feeling is that you'll need to contact the provider.
The other thing that comes to mind is that your PRI card is having
If I read Kevin's post correctly, his statement applies to ALL echo cancellers,
not just software EC.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo Santos
Sent: Wednesday, September 14, 2011 10:52
On 09/14/2011 10:14 AM, Eric Wieling wrote:
If I read Kevin's post correctly, his statement applies to ALL echo cancellers,
not just software EC.
That's correct; regardless of whether the EC runs on a DSP or on the
host CPU (or is a device in the path of an T1/E1 circuit), if it can
only
On 09/14/2011 02:37 AM, Lee, John (Sydney) wrote:
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
I'm trying to simulate the situation:
SIP Asterisk --- PSTN
In this case 16 ms works?
I've read in voip-info: Simplistically, you'd need a tail circuit (the
distance between your echo canceller and the source of the echo) of over
2500 miles to acheive an echo path of 30ms [...]
Hi,
I just wanted to clear a doubt I had. In a SIP trunk, will it show OK
status even if only one side of the SIP trunk is configured when we do sip
show peers ??
If yes, is there any other way to make sure that the trunk is ready for
making calls?? [?]
Last day we had a situation here at my
If you use qualify=yes, you should only get OK when the line is functional.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm
Sent: Wednesday, September 14, 2011 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Oh.. thank you. That could be the reason. Let me try that.
But In fact, I thought qualify = yes is used to send some thing like *keep
alive* packets in an already connected trunk to make sure the trunk is still
alive.
In my case the trunk was completely down, and then it was showing status OK
as
KA packets is one (perhaps the stated) function. But, in my experience, you
can run a dead SIP line with qualify=no.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm
Sent: Wednesday, September 14, 2011 4:18 PM
To: Asterisk Users
On 09/14/2011 02:37 PM, Gustavo Santos wrote:
I'm trying to simulate the situation:
SIP Asterisk --- PSTN
In this case 16 ms works?
I've read in voip-info: Simplistically, you'd need a tail circuit
(the distance between your echo canceller and the source of the echo) of
over 2500
chan_sip does not support specification of the password to be used for
authentication in the dial string itself;
your :password suffix is just being sent to the target system and it
is trying to find a matching extension in the dialplan (and failing).
Thanks Kevin. This is what I reckon from
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