יעע
-Original Message-
From: Vieri
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 17 Apr 2012 23:27:10
To:
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] hints and server-side DND (do not disturb)
Hi,
Currently I'm using
Hi,
Currently I'm using hints to determine SIP presence. As I understand it, a SIP
extension can be labeled as busy, ringing, etc, based on a channel status. So a
channel MUST be present. If it isn't then the extension is considered to be
"available".
If my statement is correct then is there a
- Original Message -
> From: "Yaroslav Panych"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, April 17, 2012 6:56:17 PM
> Subject: Re: [asterisk-users] Incoming SIP call is rejected always.
>
> 2012/4/18 Matthew Jordan :
> > I imagine that this is t
2012/4/18 Matthew Jordan :
> I imagine that this is the case, as ASTERISK-19601 noted that
> when this situation occurs, the NOTICE message indicates that
> there is a failure to match the extension, as opposed to a failure
> to match an allowed domain.
Yes, it was hell to detect real error cause
Without knowing the URI the INVITE request was addressed to, its
difficult to say what might be the actual cause of this. However,
in your SIP configuration you have set allowexternaldomains to no.
That implies that if the domain of the URI does not match any
of the allowed domains you have set, t
On 04/17/2012 06:17 AM, Larry Moore wrote:
The send log you have posted does not show any outgoing T.38 packets
from your system.
I set up a test build of 1.8.11.0 using the patch recently released, I
have difficulties sending T.38 with this patch, in fact I cannot send
successfully however I ca
Hello All,
I'm gettint this error, started recently when I upgraded to 1.8.10 from
1.8.4.
[Apr 17 08:03:52] ERROR[9099]: netsock2.c:263 ast_sockaddr_resolve:
getaddrinfo("external out", "(null)", ...): Name or service not known
[Apr 17 08:03:52] WARNING[9099]: chan_sip.c:26503 sip_request_call
2012/4/17 Danny Nicholas :
> Maybe it needs to be _4001020?
>
Not, it doesn't. Actually I have traced this incoming call step by
step. Real reason it refuses - wrong domain. But why it wrong - have
not any idea.
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Billy,
I really should have had my coffee before answering you previous
message. My head was in the wrong place (not saying where) and I sent
you down the wrong path.
Macro() is not the answer because of the WaitExten(). When WaitExten is
used in a Macro(), it does not match within the mac
Maybe it needs to be _4001020?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav
Panych
Sent: Tuesday, April 17, 2012 7:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk
On 04/16/2012 04:09 PM, Larry Moore wrote:
I have experienced this issue with a provider with Asterisk 1.2, 1.6 & 1.8.
I never got to the root cause of the problem however it used to occur
quite frequently, now it appear to occur once every month or two -
haven't seen it occur for a while now bu
I think OP wants DISA input sent to MYSQL, so it seems to me that an AGI
would be more appropriate. The AGI would read, do DISA, call and record the
result to the CDR without the "Ugly" dialplan SQL stuff.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-u
On Tuesday 17 Apr 2012, cjwstudios wrote:
> Looking for quotes on a very simple script that will require a pin
> number before allowing a call to be placed. The pin number would be
> recorded to their mysql CDR. Thank you.
Will the DISA application do what you need?
Regards,
-- Raj
--
Raj Mat
I'm trying to find if a realtime queue member is paused or not from the
dialplan.
For a "paused", "not in use" phone, DEVICE_STATE returns "not in use" only. Is
there a function that will tell if the phone is paused or not (other than
querying the database directly)?
Thanks,
Matt
Looking for quotes on a very simple script that will require a pin
number before allowing a call to be placed. The pin number would be
recorded to their mysql CDR. Thank you.
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On Tue, 17 Apr 2012, Bryant Zimmerman wrote:
example: l_databaseVariableName = MyTrunk
l_databaseVariableValue = ${myglobalvar}
myglobalvar = Target_Trunk
exten => doVtype-1,1,Set(${l_databaseVariableName}=${l_databaseVariableValue}
I need variable MyTrunk to = Target_Trunk
The above sets MyTr
I did this in 1.4 using hints. The most efficient (IMO) approach now would
probably to use "core show channels verbose".
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Tuesday, April 17, 2012 5:29
I have a string value from a database that has a reference to a global
variable ${myglobalvar}. When I set the value it sets it to
the string what is in the database and does not evulate the variable
inside. Any ideas how to force an evaluation as part of a set?
example: l_databaseVariableNa
Greetings Dale,
Thanks for the help I have updated my file to include the macro sample you
gave me.
The system can make the recordings once I daily the required extension in
this case 3552
--- config section for 3552 -
exten => 3552,1,Macro(timo,contentdb)
exten => 3552,n,Hangup()
---Below is
Hi
Have an asterisk. Setup a couple of friends.
Sip.conf - http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR
NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from
'RMT20' (192.168.8.1:5062) to extension '4001020' reje
The send log you have posted does not show any outgoing T.38 packets
from your system.
I set up a test build of 1.8.11.0 using the patch recently released, I
have difficulties sending T.38 with this patch, in fact I cannot send
successfully however I can receive. I did however observe some out
On 04/16/2012 04:09 PM, Billy Kaye wrote:
Thanks Dale,
Am not sure why it was working in 1.4 but for some reason it was (
Note : My Asterisk is running bundled with Elastix).
But any your suggestion worked very fine.
Glad to hear it.
Now am having one problem how can define those extension
Il 17/04/2012 01:10, Niccolò Belli ha scritto:
Tomorrow I will try without directmedia=yes.
Unfortunately it didn't help.
Niccolò
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New to Asterisk? Joi
Hi,
I'm wondering if someone has already done a web application that queries
'ExtensionStatus' events.
On my web site I have an extension listing. Next to each number I'd like to add
an icon or something that shows the extension status. I'd like this status to
be as real-time as possible. Bein
16 apr 2012 kl. 15:31 skrev Matthew Jordan:
> It's not a bug - decrementing the CSeq header field value is directly in
> violation of RFC 3261. From section 22.2:
>
> When a UAC resubmits a request with its credentials after receiving a
> 401 (Unauthorized) or 407 (Proxy Authentication Requ
To answer my own question: I stumpled upon the RetryDial function. This is
exactly what I needed! But when the function played back the audio, I
couldn't hear it. I needed to open the channel, which the Dial command
would normally do automatically. So I came up with this dialplan:
exten => 120,hin
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