Did you copy the asterisk-mib.txt and digium-mib.txt to the proper folder on
your distro?
I see people forgetting about that step.
On Wed, Sep 14, 2011 at 2:18 AM, Ishfaq Malik wrote:
> Hi
>
> I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as
> my resource
>
> http://ofps
On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
Please let me know the correct procedure to get .alaw file format since
I belong to India region.
Well, let's see...
You used '-t ul' and got a 'ulaw.'
I wonder what '-t al' will give you?
Failing that, I suspect 'sox --help' or Google would be a
> 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 fr
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 25
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 Us
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 s
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
S
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
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 [...] A
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]
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 han
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 AM
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 issues.
So any software echo canceller available in dahdi isn't good enough?
2011/9/13 Kevin P. Fleming
> 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 configuration:
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 => s,n,Background(1-ok
+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 Noll
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
-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: authenticatio
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 D-chann
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 => s,n,MYSQL(D
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 wrote:
> **
> Hello,
>
> I do the following in a macro in the dialpl
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 ${connid})
But nothing changes in m
On Wed, Sep 14, 2011 at 4:08 AM, Steve Edwards wrote:
> On Wed, 14 Sep 2011, Israel Gottlieb wrote:
>
> is it possible to pas variables to the shell function
>>
>> Set(recordingavail=${SHELL("ls /var/lib/asterisk/sounds/**
>> custom/${TOPMENU}")})
>>
>> im trying to see if a file is available bef
On Wed, Sep 14, 2011 at 5:27 AM, Dale Noll wrote:
> On 09/13/2011 07:49 PM, Israel Gottlieb wrote:
>
>> is it possible to pas variables to the shell function
>>
>> Set(recordingavail=${SHELL("ls
>> /var/lib/asterisk/sounds/**custom/${TOPMENU}")})
>>
>> im trying to see if a file is available befo
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 th
nitor("SIP/10001-b7d71bd0",
"/var/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/
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
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 ou
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
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 :
> Hi ,
>
> What was the solution of that problem ? Did provider change the setting at
> there end or else ?
>
> On Tue
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),
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
.1.3.6.1.4.1
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
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 g
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 plac
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 a
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