That's really good info Tony!
Thanks very much for the response!
I will consider this to implement a better approach for the failed cases!
Cheers,
Patrick Wakano
On 14 March 2018 at 20:44, Tony Mountifield wrote:
> In article t...@mail.gmail.com>,
> Patrick Wakano wrote:
> >
> > Thanks Dovid!
In article ,
Patrick Wakano wrote:
>
> 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 wa
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.or
I would think that is a bug since the only time DIALSTATUS = BUSY is where
you got a 486 or 600 (as per
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings).
On Tue, Mar 13, 2018 at 10:11 PM, Patrick Wakano wrote:
> Hello list,
> Hope all doing well!
>
> I've been checking some case
Hello list,
Hope all doing well!
I've been checking some cases when a Dial fails and dialplan execution
continues to handle this. I am finding it a little confusing how we should
handle the DIALSTATUS and the HANGUPCAUSE in this situation
More specifically, I am facing a case in version 13.6.0
On Tue, Dec 22, 2015 at 7:26 AM, Marcos Prates
wrote:
> Hi,
>
> I'm having a strange problem with Asterisk 13 i can't seem to find out
> whats causing it.
> After a Dial call from one SIP peer to another, if the calling side hangs
> up, DIALSTATUS is not set, but when the called side hangs up, it
Hi,
I'm having a strange problem with Asterisk 13 i can't seem to find out whats
causing it.
After a Dial call from one SIP peer to another, if the calling side hangs up,
DIALSTATUS is not set, but when the called side hangs up, it does.
The strangest thing is when debugging SIP, it sends/rec
This works fine for me,
$dialstatus = $agi->get_variable("DIALSTATUS");
$cdr['dialstatus'] = $dialstatus['data'];
Try as it is, I believe it's because of concatenation.
Regards,
Zohair Raza
On Fri, Dec 2, 2011 at 4:27 PM, Tony Mountifield wrote:
> In article
Can anybody please reply on this?
Regards,
Kamlesh
From: kamlesh_...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 27 Dec 2011 09:49:21 +
Subject: Re: [asterisk-users] DIALSTATUS Values
Hello,
After investing some time, I could come to know the reason for not
K | cut -f 1 -d
/ | grep '100' ")
Could you please suggest now how to rectify this?
Regards,
Kamlesh
> To: asterisk-users@lists.digium.com
> From: t...@softins.co.uk
> Date: Fri, 2 Dec 2011 12:27:19 +
> Subject: Re: [asterisk-users] DIALSTATUS Values
>
In article ,
Kamlesh Kumar wrote:
> In addition to my reply:
>
> I used to fetch the value using print_r function but that also tells that
> there is no value
> in data section.
> $dialstatus=$agi->get_variable(DIALSTATUS);
> print_r($dialstatus);
>
> AGI Rx << GET VARIABLE DIALSTATUS
> AGI T
[Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << [data] =>
Regards,
Kamlesh
From: kamlesh_...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] DIALSTATUS Values
Date: Fri, 2 Dec 2011 11:58:26 +
bose("Status".$dd);
AGI Rx << GET VARIABLE DIALSTATUS
AGI Tx >> 200 result=1 (CANCEL)
AGI Rx << VERBOSE "Status" 1
Regards,
Kamlesh
> To: asterisk-users@lists.digium.com
> From: t...@softins.co.uk
> Date: Fri, 2 Dec 2011 11:44:34 +
> Subje
In article ,
Kamlesh Kumar wrote:
> I tried to search the answer of my problem but unable to get resolution so
> sending to you
> guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm
> unable to
> retrieve the DIALSTATUS value, during execution of AGI script, I get empty
1
AGI Rx << GET VARIABLE DIALSTATUS
AGI Tx >> 200 result=1 (ANSWER)
AGI Rx << VERBOSE "Status" 1
AGI Tx >> 200 result=1
Regards,
Kamlesh
Date: Fri, 2 Dec 2011 16:26:50 +0500
From: govoi...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [ast
0X.,n,AGI(isdcall.php)
>
> Regards,
> Kamlesh
>
> --
> Date: Fri, 2 Dec 2011 16:16:27 +0500
> From: govoi...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] DIALSTATUS Values
>
>
> Hi,
> How are you callin
Hello,
in /etc/extension.conf
[privoip]
exten => _00X.,n,AGI(isdcall.php)
Regards,
Kamlesh
Date: Fri, 2 Dec 2011 16:16:27 +0500
From: govoi...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS Values
Hi,
How are you calling this AGI in your dialp
Hi,
How are you calling this AGI in your dialplan !!?
Regards,
Sammy.
On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar wrote:
> Hello,
>
> I tried to search the answer of my problem but unable to get resolution so
> sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts
> usin
Hello,
I tried to search the answer of my problem but unable to get resolution so
sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts
using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI
script, I get empty value.
Extracts from AGI Script:
o CDR translator in testing
and this is looking very promising.
Thanks for your help.
Bryant
From: brya...@zktech.com
Sent: Friday, December 24, 2010 9:28 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users
t would make sense to set the inital state of the
>>>> ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
>>>> I may be missing the point on this can anyone else speak to it?
>>>>
>>>> Bryant
>>>>
>>>
an Harutyunyan"
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN => {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h => {
Noop(${DIALSTATUS}
t;>> SIP/userN-b6317738
>>> == Spawn extension (fu, 00018185402020, 2) exited non-zero on
>>> 'SIP/user3-b6317738'
>>> -- Executing [...@fu:1] NoOp("SIP/userN-b6317738", "CANCEL") in new stack
>>>
>>> I think, I am right
>>>
>>> --
>>> Vardan Har
the
>> ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
>> I may be missing the point on this can anyone else speak to it?
>>
>> Bryant
>>
>> --------
>> *From*: "Vardan Harutyunyan"
>> *Sent*: Thursday, December 23, 2010 2:
yone else speak to it?
Bryant
*From*: "Vardan Harutyunyan"
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context f
one else speak to it?
Bryant
From: "Vardan Harutyunyan"
Sent: Thursday, December 23, 2010 2:11 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN => {
Thanks Vardan,
You're right. Running the script under h extension gets me the results I'm
looking for.
On Wed, Dec 22, 2010 at 5:38 PM, Vardan Harutyunyan wrote:
> Try to use h extension
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep E
;
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
The Dial Status is not set when accessing it from the h extension.
Bryant
From: "Vardan Harutyunyan"
Sent: Wednesday, December 22, 2010 10:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
Tr
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are
g as
well.
Bryant
From: "Michael"
Sent: Wednesday, December 22, 2010 9:42 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
Hi Nikhil,
Both debug and verbose are set t
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can see,
for the other types of reasons, the DIALSTATUS got a value (and we see the
events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil wrote:
> Hi
>Enable debug level to more tha
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero on
'SIP/Proxy-0031'
--
_
-- Bandwidth a
Anyone??
Thanks.
On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question wrote:
> Hello,
>
> We have a strange situation (asterisk 1.6.2.14), where we get a result for
> DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
>
> This is the (relevant) test dialplan:
> ---
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTA
Hi,
Which asterisk version are you using. try setting call-limit value in
sip.conf and see if it makes any difference.
On Thu, Oct 21, 2010 at 1:29 PM, GBR Icasiano, Ryan A. <
raicasi...@globalbridgeresources.com> wrote:
> Hi,
>
> Here is the scenario:
> 1. 1st phone calls and asterisk dials
Maybe you should post this portion for your dialplan. I have done the same
thing several times and never had this timeout issue.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 4:08 AM, "GBR Icasiano, Ryan A." <
raicasi...@globalbridgeresources.com> wrote:
Hi,
Here is the scenario:
1. 1
Hi,
Here is the scenario:
1. 1st phone calls and asterisk dials to extension no.
2. Extension answers 1st caller(which makes it busy).
2. 2nd phone calls and asterisk dials to extension no.
3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in
DIAL cmd) before proceeding t
Hi!
> > could anybody tell me if the info below is still correct:
> >
> > Note: In order to obtain useful DIALSTATUS information when dialing a
> > peer you will need to have qualify=yes in that peer's definition (e.g.
> > in sip.conf or iax.conf).
> > http://www.voip-info.org/wiki/view/Asterisk+
On Sat, 2010-04-17 at 17:38 +0400, Rustam Kovhaev wrote:
> could anybody tell me if the info below is still correct:
>
> Note: In order to obtain useful DIALSTATUS information when dialing a
> peer you will need to have qualify=yes in that peer's definition (e.g.
> in sip.conf or iax.conf).
> http
Hi there,
could anybody tell me if the info below is still correct:
Note: In order to obtain useful DIALSTATUS information when dialing a
peer you will need to have qualify=yes in that peer's definition (e.g.
in sip.conf or iax.conf).
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATU
On 11/02/09 07:28, Steve Edwards wrote:
>On Mon, 2 Nov 2009, Patrick Plattes wrote:
>
>> you can do print the dialstatus to the console e.g.:
>> exten => s,n,NoOp(${DIALSTATUS})
>>
>> More info:
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
>
>A "better practice" would be to use verbose()
On Mon, 2 Nov 2009, Patrick Plattes wrote:
> you can do print the dialstatus to the console e.g.:
> exten => s,n,NoOp(${DIALSTATUS})
>
> More info:
> http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
A "better practice" would be to use verbose() -- an application with
greater functionality wr
Hi,
you can do print the dialstatus to the console e.g.:
exten => s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
Bye,
Patrick
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
I can not seem to get dial status to work,
in sip.conf I have: qualify=yes
simple plan:
exten => 51,1,Dial(SIP/11,20,r)
exten => 51,n,Goto(s-${DIALSTATUS},1)
exten => s-Busy,1,Hangup()
exten => s-Answer,1,Macro(atb)
I'm dialing from exten.11 to exten.11 so I get busy signal and the channel
shoul
My call file was calling an AGI application, and from with the AGI, I could
not get the DIALSTATUS, I will try to send it to the dialplan first, then
call my AGI from the dialplan and see what happen.
Thanks for your help
On Tue, Feb 3, 2009 at 3:35 AM, Johansson Olle E wrote:
>
> 3 feb 2009
3 feb 2009 kl. 04.33 skrev Ex Vito:
> On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno
> wrote:
>> Is it possible to retrieve the DIALSTATUS variable when placing
>> call through
>> a call file. This variable is set when using the Dial()
>> application from
>> the dialplan, but I am using a
On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno wrote:
> Is it possible to retrieve the DIALSTATUS variable when placing call through
> a call file. This variable is set when using the Dial() application from
> the dialplan, but I am using a call file for my current application and need
> to get t
Hello,
Is it possible to retrieve the DIALSTATUS variable when placing call through
a call file. This variable is set when using the Dial() application from
the dialplan, but I am using a call file for my current application and need
to get the dialstatus.
Thank you.
_
--- Matt Riddell <[EMAIL PROTECTED]> wrote:
> http://bugs.digium.com/view.php?id=12230
Thanks Matt.
However, "I may be wrong" but this isn't exactly what
I'm looking for. I would like Asterisk to
"transparently" set my CDR(disposition) field to
reflect if a call has simply timed out (NO ANSWER)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Vieri wrote:
> --- Ex Vito <[EMAIL PROTECTED]> wrote:
>
>> ...as long as the destination does not answer
>> you'll get
>> a NO ANSWER disposition.
>> So, if in your case you want to know if a user
>> answered
>> the phone, then, yes, you will
--- Ex Vito <[EMAIL PROTECTED]> wrote:
> ...as long as the destination does not answer
> you'll get
> a NO ANSWER disposition.
> So, if in your case you want to know if a user
> answered
> the phone, then, yes, you will have to add the
> DIALSTATUS
> value to the CDR, probably in the CD
...as long as the destination does not answer you'll get
a NO ANSWER disposition.
Note, however, that "answering" can be one of:
- Dial a phone and the user answers the phone
- Connecting the caller to voicemail, for example,
after Dial timed out
- Playing an IVR / sound / music
--- Vieri <[EMAIL PROTECTED]> wrote:
> 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
>
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 abl
Oh, for god's sake.
how stupid is I am feeling :)
My brain cell is feeling very ashamed.
Julian.
James FitzGibbon wrote:
> On 8/3/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
>> why if I call the Busy or Congestion extensions, the DIALSTATUS and
>> HANGUPCAUSE variables are not set ?
>>
>
On Fri, 2007-08-03 at 19:58 +0100, Julian Lyndon-Smith wrote:
> why if I call the Busy or Congestion extensions, the DIALSTATUS and
> HANGUPCAUSE variables are not set ?
The DIALSTATUS channel variable is set when you call the Dial()
application. If you don't call the Dial() application (like if
On 8/3/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
>
> why if I call the Busy or Congestion extensions, the DIALSTATUS and
> HANGUPCAUSE variables are not set ?
>
> If I call (say) extension 1234 all things are set ok.
I think you've answered your own question there. The only asterisk
app
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(${TEST})
exten => _X.,n,Dial(SIP/${EXTE
exten => s,2,Goto(s-${DIALSTATUS})
ref:
http://www.voip-info.org/wiki/view/DIALSTATUS
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
I also use HANGUPCAUSE in some circumstances.
exten => s,2,Goto(s-${HANGUPCAUSE})
ref:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Variable+HAN
Check out this example dialplan: http://pastebin.ca/19456
That should give you everything you need.
bp
On 6/6/06, Moises Silva <[EMAIL PROTECTED]> wrote:
this is what I have, and it works on Asterisk-1.2.1[macro-sipextens]exten => s,1,Macro(validate_extension)
exten => s,2,Dial(SIP/${sipprefix}$
this is what I have, and it works on Asterisk-1.2.1
[macro-sipextens]
exten => s,1,Macro(validate_extension)
exten => s,2,Dial(SIP/${sipprefix}${num},${calltimeout}|${calloptions})
exten => s,3,Macro(catch_dial_response,${DIALSTATUS})
so, After Dial, I catch the dial response, and heres the catc
I tried with CHANUNAVAIL but I was not successful. I want to try to call a
SIP client. If it is not answering and cannot be found I want wo call
someone else.
How can I do that? NOANSWER and CHANUNAVAIL do not work out.
> Wether the SIP client is not registered or does not exists at all you
> will
Wether the SIP client is not registered or does not exists at all you
will get CHANUNAVAIL.
Regards
On 6/6/06, Christophorus Laube <[EMAIL PROTECTED]> wrote:
Hi,
I use an E1-Board to hand the calls over to internal SIP-Clients. My
Question is which Dialstatus is set when the SIP-client is unre
Hi,
I use an E1-Board to hand the calls over to internal SIP-Clients. My
Question is which Dialstatus is set when the SIP-client is unreachable.
I tried with NOANSWER but does not seem to be suitable.
Does anyone of you have a solution?
In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but t
Hi all,
i just have a question: could i Known the state of a
SIP phone without make it a Dial ?
Thanks
Giordano
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options
Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers
>
> When you pass several Dial strings only the last exited
> channel DIALSTATUS is saved. In the case that 1 of the
> channels answer, the status will be ANSWER obvi
> >>Douglas Garstang
> >>Sent: Friday, April 07, 2006 2:21 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers
> >>
> >>Folks,
> >>
> >>When I
Variable name to track each one.
>>-Original Message-
>>From: [EMAIL PROTECTED]
>>[mailto:[EMAIL PROTECTED] On Behalf Of
>>Douglas Garstang
>>Sent: Friday, April 07, 2006 2:21 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subjec
Folks,
When I have a dial string like this:
Dial(SIP/3254101&SIP/3254102,20,tr)
and I want to check the ${DIALSTATUS} variable after the dial, how do I know
which number I am getting the variable for?
And, what about this?
Dial(SIP/3254101&SIP/[EMAIL PROTECTED],20,tr)
What happens in that ca
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
Code Lover wrote:
Hi all,
How i can call my perl agi script when the call is hungup. Because i
am making some external Cdr calculation.
Hi M. Lover,
There are two solutions for you:
- You can call an AGI on hangup by using the extension 'h' : exten =>
h,1,DeadAGI(myagi.agi)
- If you're us
Hi all,
How i can call my perl agi script when the call is hungup. Because i
am making some external Cdr calculation.
--
Best Regards,
Abdul Lateef Khan
Computer Programmer
Mobile No. : +974 - 5405022
ICQ : 276-994-704
YM! : [EMAIL PROTECTED]
MSN : [EMAIL PROTECTED]
Google Talk : [EMAIL PROTECTED]
Hi all,
I would like to run my perl agi script when the call is hungup. I did
one script to calculate calling balance and duration.
I made one timer Where the dialstaus is Answered But i am am in
confiuse how i can stop my timer when the dialstus will be hangup.
Please give me an advice to solve
On Wed, 2005-09-21 at 14:36 +1000, Mark Edwards wrote:
> terminating asterices. (Is that the plural of asterisk?)
I propose asterii, while by no means gramatically correct it wont fall
under potential sue happy lawyers who own the unix trademark (after all
the plural there is unices). oh no I sa
Have come to a solution on this, and as I suspected, the issue appears
to be a bit of a version mismatch between terminating asterices. (Is
that the plural of asterisk?) Anyway, to cut a long story short, I
tested with another provider, found that they were running a later
version (nearer CVS-HEAD)
I met same problem when dial via zap channel.
Does anyone know how to solve it?
thanks.
2005/9/15, Mark Edwards <[EMAIL PROTECTED]>:
> Hi.
>
> I'm dialling two numbers - one that's unobtainable, one that's busy.
>
> ${DIALSTATUS} is coming back ANSWER each time right before the channels hang
>
I'm not expert on this matter,but base on experience that is a normal situation on SIP/IAX channels since the call made is answered by the other end regardless of the situation you might found.
I'm on PRI ISDN and for me dialstatus and hangupcause works pretty good.
Regards
Jb
On 9/15/05, Mark Edw
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds
On Sun, 2005-08-28 at 12:45 -0700, Geoff Karl wrote:
> If you are using Async and the action ID for some reason the Event:
> Newstate doesn't respond with the ActionID, but only a automatically
> generated "Uniqueid".
When using Async you receive an OriginateSuccess or OriginateFailure
event.
Thes
On 28 Aug 2005 10:35:34 -, saket setu <[EMAIL PROTECTED]> wrote:
>
>
>
> Hi all,
> I am from India and has been recently using asterisk for testing and
> enahncing my telephony knowledge. I am trying to use the originate Command
> from the Asterisk manager on both SIP and ZAP. The co
Hi all,
I am sending the mail again as there was some mistake in the dial plan in the last mail send:
I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. T
Hi all,
I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER a
Manuel Schroeder wrote:
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
__
> I'm having an issue with my current configuration. I have a single
> PSTN line connected to an X100P and a couple IAX trunks to NuFone and
> VoipJet. When I make an outbound call it doesn't properly detect if
> my PSTN line is in use with another call and then overflow to my
> outbound IAX conn
I'm having an issue with my current configuration. I have a single
PSTN line connected to an X100P and a couple IAX trunks to NuFone and
VoipJet. When I make an outbound call it doesn't properly detect if
my PSTN line is in use with another call and then overflow to my
outbound IAX connections.
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
On December 4, 2004 08:43 am, Rich Adamson wrote:
> Looks like the command is documented in the current config samples.
Yeah I see that now. :-)
> Since the comments use words like "doesn't work with all telcos",
> could this have something to do with detecting busy when a call
> reaches a desti
On Sat, 4 Dec 2004, Rich Adamson wrote:
> > The mind boggles -- PRI is *always* out of band.
>
> Looks like the command is documented in the current config samples.
>
> I'm not knowledgable/experienced (as yet) on where it is actually used,
> but just reading the comments in the config sample le
> On December 3, 2004 03:36 pm, Andrew Kohlsmith wrote:
> > exten => 1234,1,Dial(Zap/g1/5551234,,g)
> > exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
> > ${DIALSTATUS})
> >
> > Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not
> > be BUSY?
>
> Brian W
On December 3, 2004 03:36 pm, Andrew Kohlsmith wrote:
> exten => 1234,1,Dial(Zap/g1/5551234,,g)
> exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
> ${DIALSTATUS})
>
> Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not
> be BUSY?
Brian West pointed me at
Just throwing this out here, hopefully someone can tell me why.
*CLI> show version
Asterisk CVS-HEAD-11/17/04-10:16:38 built by [EMAIL PROTECTED] on a i686
running
Linux
Zap/g1 is pri_cpe to Bell Canada
5551234 is a normal POTS line I have busied out (handset offhook)
exten => 1234,1,Dial(Zap/
Hello All,
Just a very simple example. I'm trying to make a call to
a busy phone number using Dial application.
-- H.323 call to 12345 with codec ALAW
-- Called 12345
-- OH323/L5663 is ringing
-- H.323 call 'ip$localhost/5663' cleared, reason 18
(Remote endpoint is bus
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