Displaying 20 results from an estimated 3000 matches similar to: "get dialstatus variable when returning No such context/extension"
2009 Jan 05
0
G729 VAD issue
Hi,
My setup is SIP Call-->Asterisk-->VSP1 or VSP2 or VSP3
I'm experiencing an interconnect issue with one of the VSP's that seems to
have to do with Asterisk not having any VAD control. The error is:
NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of
G.729 since we already have a VAD frame at the end
The VSP has switched off silence suppression on their
2009 Jan 06
0
G.729 VAD issue
Hi,
My setup is SIP Call-->Asterisk-->VSP1 or VSP2 or VSP3
I'm experiencing an interconnect issue with one of the VSP's that seems to
have to do with Asterisk not having any VAD control. The error is:
NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of
G.729 since we already have a VAD frame at the end
The VSP has switched off silence suppression on
2007 Feb 27
1
Not registering Port with VSP
Hello All,
For some reason my asterisk server is not registering a port number with
my VSPs. This is causing problems where people are not able to dial in
from any of my SIP or IAX VSPs.
I do have one VSP that has hard coded my IP and port so I can get
incoming calls but this still leaves a problem with my other VSPs.
Hose can I get asterisk to register my IP and port? I have been
2007 Jan 08
0
Allowing inbound VoIP Calls from VSP
Hi All,
I think I have missed something as I am resisted with 4 VSPs and I can
not dial in using any one of them using the corresponding VoIP numbers
assigned with the VSP. I can make outbound calls to another VoIP number
to the same provider.
The weird thing is that I have a DID with a VSP and I have that working
fine but try using the associate VoIP number and nothing happens.
When
2007 Jan 12
1
Not Registering Port with VSP.
Hi All,
I seem to be having a problem with all my VSPs. When I am registering
with them I don't seem to be passing my port number. This problem
causes other users the inability to call my VoIP number with the VSP.
My VSP showed me what they are seeing.
I have changed my useragent to be: Linksys/SPA941-4.1.15
Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no
2005 Sep 15
0
dialplan to try VOIP providers if they can't terminate call
I am trying to figure out how to try different VOIP providers if they
aren't able to terminate the call because they don't offer service to
that dialing area.
The error that gets logged to the console is:
Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call
rejected by x.x.x.x: No such context/extension
The dialstatus returned is "No Answer"
I could check for
2006 Jan 21
0
Dialstatus Oddity in 1.2
Hello all,
I am working on a creating some intelligent failover dial-plan
logic and I'm running into something that I'd like some feedback on.
Basically, it appears that if you place a call to an IAX2 peer that
refuses the connection, or is unavailable, a NOANSWER dialstatus is
returned.
Example:
-- Executing Macro("IAX2/cubix-19",
2004 Dec 03
2
DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
Just throwing this out here, hopefully someone can tell me why.
*CLI> show version
Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running
Linux
Zap/g1 is pri_cpe to Bell Canada
5551234 is a normal POTS line I have busied out (handset offhook)
exten => 1234,1,Dial(Zap/g1/5551234,,g)
exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
2005 Jan 21
0
Manager API on gives the DIALSTATUS of the first picked up channel?
Hi All!
Let me explain the problem. When using the Originate?
command from the manager api, the dialstatus variable returns results?
for whichever phone picks up first, and in this case it is the IAX/2?
connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,?
or an extension either. What I am ultimately trying to do is get the?
dialstatus of the Zap/X/XXXXXXX channel, i.e.,
2013 Apr 01
0
Getting DIALSTATUS via agi
Hi all,
Hopefully, I just need a second set of eyes on this one, but I just can't
figure out what I'm doing wrong. I'm using an agi script to dial a number,
check the dial result, and act accordingly.
The problem is that I'm not getting anything back from DIALSTATUS, or
HANGUPCAUSE.
Here is the relevant perl code:
===============================================================
2011 Aug 14
1
1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered)
I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. Below is a dialplan snippet and the resulting CLI output. This is running in an 'h' extension.
Noop(DIALSTATUS=${DIALSTATUS})
Noop(CDR(disposition)=${CDR(disposition)})
-- Executing [h at pbxmax-dial-simple:1] NoOp("SIP/msx_01-0000005b", "DIALSTATUS=ANSWER") in new stack
2018 Mar 14
2
DIALSTATUS vs HANGUPCAUSE
Thanks Dovid!
Indeed looks a bug but regardless of this, this problem made me think that
the HANGUPCAUSE could be used for this purpose with benefits.
I couldn't find an explanation about when DIALSTATUS would actually be
better.
The HANGUPCAUSE was reworked in version 11 (
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find
someone actually stating it is a better
2004 Dec 12
0
DIALSTATUS missing an important condition?
I have recently built my first asterisk system and am very impressed with
its capabilities.
However, I have run into one problem that hopefully someone can help me
with.
I am trying to use the DIALSTATUS function to route incoming calls to the
appropriate Voice Mail (busy or unavailable) or to an Unavailable Number
recording if the number is not assigned.
However, I find that DIALSTATUS
2013 May 05
1
GotoIf DIALSTATUS - not working
What am I doing wrong?
Goif dialstatus: busy CONGESTION not working.
exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr)
exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2)
exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr)
exten => _7NXXXXXX,n,Hangup()
When I try to
2007 Aug 03
2
DIALSTATUS not set
I'm trying to write a dialplan that will allow me to "stress" test it. I
want to be able to dial an extension, or pretend that the extension is
busy or out of order (so that I can see what to do)
given the dialplan snippet:
[outbound]
exten => _X.,1,NoOp(${TEST})
exten => _X.,n,Dial(SIP/${EXTEN})
exten => Busy,1,Busy(2)
exten => Busy,n,Hangup()
exten =>
2005 Jan 21
0
Help DIALSTATUS gives ANSWER when line is BUSY?
I'm running Asterisk CVS-v1-0-12/20/04.
I'm using PHP with Manager API Here is the code:
####################################################################
# Make call
####################################################################
$socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout);
if (!$socket) {
echo "$errstr ($errno)<br /\n";
} else {
2009 May 20
1
Macro with DIALSTATUS
Hi,
I am trying to pass DIALSTATUS to a Macro so that i can set a
variable when a call is placed (call is placed via a call file to
another extension first). Basically i don't want to dial a number
where a call is already bridged and thats why i am setting a variable.
[macro-afterdial];
exten => s,1,Goto(s-${ARG1},1)
exten => s-ANSWER,1,SetGlobalVar(NUM${ARG2} = "ACTIVE")
2008 Mar 10
2
dialstatus and cancelled calls
According to
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
when a caller hangs up before the callee has time to
pick the phone up then DIALSTATUS should be CANCEL.
And it is.
However, the disposition field in the CDR table is "NO
ANSWER".
So if I analyze the CDR data I won't be able to
discriminate calls cancelled by the caller and calls
not answered by the callee
2004 Dec 04
1
more DIALSTATUS/HANGUPSTATUS woes with IAX2
Phone - TDM430P - home* - IAX2 - office* - PRI - Telco
I dial a busy number from the Phone.
Home* shows this in the CLI:
-- Executing Macro("Zap/1-1", "dial-wu|2922004") in new stack
-- Executing Dial("Zap/1-1", "IAX2/andrew@wu-ast/2922004||g") in new stack
-- Called andrew@wu-ast/2922004
-- Call accepted by wu-ast (format gsm)
--
2013 Jul 03
1
SIP. Call-limit dialstatus
Hi all. We have a problem with correct dialstatus and cdr(disposition) when
using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and
CDR(disposition)='NO ANSWER'
-- Executing [0014 at sub_pbxdialco:49] Dial("SIP/1295-000001f8",
"SIP/0014,12,tTkK") in new stack
== Using SIP RTP CoS mark 5
[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003