Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE

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

Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE

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

Re: [asterisk-users] 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 (

Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE

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

[asterisk-users] DIALSTATUS vs HANGUPCAUSE

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

Re: [asterisk-users] DIALSTATUS not being set

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

[asterisk-users] DIALSTATUS not being set

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

Re: [asterisk-users] DIALSTATUS Values

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

Re: [asterisk-users] DIALSTATUS Values

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 t...@softins.co.uk

Re: [asterisk-users] DIALSTATUS Values

/ | 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 snt142-w54267269808afd17bccd5891...@phx.gbl

[asterisk-users] DIALSTATUS Values

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:

Re: [asterisk-users] DIALSTATUS Values

Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: 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

Re: [asterisk-users] DIALSTATUS Values

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 dialplan

Re: [asterisk-users] DIALSTATUS Values

(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 dialplan !!? Regards, Sammy

Re: [asterisk-users] DIALSTATUS Values

DIALSTATUS SIP/10036-00a8AGI Tx 200 result=1 (ANSWER) SIP/10036-00a8AGI Rx VERBOSE Status 1 SIP/10036-00a8AGI 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: [asterisk-users] DIALSTATUS

Re: [asterisk-users] DIALSTATUS Values

In article snt142-w45a64e4743de653da591...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com 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

Re: [asterisk-users] DIALSTATUS Values

DIALSTATUS SIP/10036-00b6AGI Tx 200 result=1 (CANCEL) SIP/10036-00b6AGI 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 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142

Re: [asterisk-users] DIALSTATUS Values

@lists.digium.com Subject: RE: [asterisk-users] DIALSTATUS Values Date: Fri, 2 Dec 2011 11:58:26 + I believe the syntax is correct because, If I use $dd=$dialstatus[code]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b2AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b2AGI Tx 200

Re: [asterisk-users] DIALSTATUS Values

In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com 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);

Re: [asterisk-users] DIALSTATUS on CANCEL

: [asterisk-users] DIALSTATUS on CANCEL If a call is hung up before an answer our h extension is not running in our dial macro Bryant On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com wrote: Hello Bryant Extension h is worked in any case of hangup. It not important

Re: [asterisk-users] DIALSTATUS on CANCEL

*From*: Vardan Harutyunyan hvarda...@gmail.com *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

Re: [asterisk-users] DIALSTATUS on CANCEL

...@gmail.com *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

Re: [asterisk-users] DIALSTATUS on CANCEL

*From*: Vardan Harutyunyan hvarda...@gmail.com *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

Re: [asterisk-users] DIALSTATUS on CANCEL

*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}); }; }; And look CLI

Re: [asterisk-users] DIALSTATUS on CANCEL

*From*: Vardan Harutyunyan hvarda...@gmail.com *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

Re: [asterisk-users] DIALSTATUS on CANCEL

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 hvarda...@gmail.comwrote: Try to use h extension -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation

Re: [asterisk-users] DIALSTATUS on CANCEL

From: Vardan Harutyunyan hvarda...@gmail.com 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

Re: [asterisk-users] DIALSTATUS on CANCEL

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 d.nik...@cem-solutions.net wrote: Hi

Re: [asterisk-users] DIALSTATUS on CANCEL

. Bryant From: Michael voip.quest...@gmail.com Sent: Wednesday, December 22, 2010 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS on CANCEL Hi Nikhil, Both debug

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 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

Re: [asterisk-users] DIALSTATUS on CANCEL

The Dial Status is not set when accessing it from the h extension. Bryant From: Vardan Harutyunyan hvarda...@gmail.com Sent: Wednesday, December 22, 2010 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS on CANCEL

Re: [asterisk-users] DIALSTATUS on CANCEL

*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 Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com

Re: [asterisk-users] DIALSTATUS on CANCEL

Anyone?? Thanks. On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question voip.quest...@gmail.comwrote: 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:

Re: [asterisk-users] DIALSTATUS on CANCEL

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

[asterisk-users] DIALSTATUS on CANCEL

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,

[asterisk-users] DIALSTATUS always returns NOANSWER

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

Re: [asterisk-users] DIALSTATUS always returns NOANSWER

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. 1st

Re: [asterisk-users] DIALSTATUS always returns NOANSWER

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 to

[asterisk-users] DIALSTATUS variable and qualify=no

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).

Re: [asterisk-users] DIALSTATUS variable and qualify=no

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).

Re: [asterisk-users] DIALSTATUS variable and qualify=no

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).

Re: [asterisk-users] Dialstatus

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 --

Re: [asterisk-users] Dialstatus

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 written

Re: [asterisk-users] Dialstatus

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() -- an

[asterisk-users] Dialstatus

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 should

Re: [asterisk-users] dialstatus through a call file

3 feb 2009 kl. 04.33 skrev Ex Vito: On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno tipas...@gmail.com 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

Re: [asterisk-users] dialstatus through a call file

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 o...@edvina.net wrote:

Re: [asterisk-users] dialstatus through a call file

On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno tipas...@gmail.com 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

[asterisk-users] dialstatus through a call file

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.

Re: [asterisk-users] dialstatus and cancelled calls

--- 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) or if

Re: [asterisk-users] dialstatus and cancelled calls

-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 have to add the

Re: [asterisk-users] dialstatus and cancelled calls

--- 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 CDR's

Re: [asterisk-users] dialstatus and cancelled calls

--- 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 NO

Re: [asterisk-users] dialstatus and cancelled calls

...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

[asterisk-users] 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

[asterisk-users] 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})

Re: [asterisk-users] DIALSTATUS not set

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

Re: [asterisk-users] DIALSTATUS 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

Re: [asterisk-users] DIALSTATUS not set

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 ? If I

[asterisk-users] DIALSTATUS and HANGUPCAUSE extensions such as s-BUSY

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:

[Asterisk-Users] Dialstatus

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

Re: [Asterisk-Users] Dialstatus

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

Re: [Asterisk-Users] Dialstatus

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

Re: [Asterisk-Users] Dialstatus

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 catch

Re: [Asterisk-Users] Dialstatus

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 =

[Asterisk-Users] Dialstatus results

Hi all, i just have a question: could i Known the state of a SIP phone without make it a Dial ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

Re: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

Subject: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers Folks, When I have a dial string like this: Dial(SIP/3254101SIP/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

RE: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

-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 obviously, but if the second fails because

[Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

Folks, When I have a dial string like this: Dial(SIP/3254101SIP/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/3254101SIP/[EMAIL PROTECTED],20,tr) What happens in that

RE: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

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 Subject: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers Folks

[Asterisk-Users] 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

[Asterisk-Users] DIALSTATUS

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

[Asterisk-Users] DIALSTATUS

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

Re: [Asterisk-Users] DIALSTATUS

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

Re: [Asterisk-Users] ${DIALSTATUS} problems

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

Re: [Asterisk-Users] ${DIALSTATUS} problems

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 up.

Re: [Asterisk-Users] ${DIALSTATUS} problems

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

Re: [Asterisk-Users] ${DIALSTATUS} problems

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

[Asterisk-Users] ${DIALSTATUS} problems

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

Re: [Asterisk-Users] DIALSTATUS for Originate

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. These

[Asterisk-Users] DIALSTATUS for Originate

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

[Asterisk-Users] DIALSTATUS for Originate Command

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.

Re: [Asterisk-Users] DIALSTATUS for Originate

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 command

[Asterisk-Users] ${DIALSTATUS}

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

Re: [Asterisk-Users] ${DIALSTATUS}

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

[Asterisk-Users] DIALSTATUS with X100P

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.

Re: [Asterisk-Users] DIALSTATUS with X100P

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

[Asterisk-Users] 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

Re: [Asterisk-Users] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)

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

Re: [Asterisk-Users] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)

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 led me

Re: [Asterisk-Users] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)

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

[Asterisk-Users] 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 [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 =

Re: [Asterisk-Users] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)

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

[Asterisk-Users] DIALSTATUS variable and oh323 channel

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