Re: [asterisk-users] DIALSTATUS on CANCEL
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the
same return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
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
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
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 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
Bryant Zimmerman wrote:
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
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 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
mailto:d.nik...@cem-solutions.net wrote:
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'
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Re: [asterisk-users] DIALSTATUS on CANCEL
In AEL macro you must use catch h
for example
macro DialToSIPProv (tech,number,prov) {
Dial(${tech}/${numb...@${prov});
switch(${DIALSTATUS}) {
case BUSY:
Noop(BUSY);
[Do some one]
break;
case CHANUNAVAIL:
Noop(CHANUN);
[Do some one]
break;
case NOANSWER:
Noop(NOANS);
[Do some one]
break;
case CANCEL:
Noop(CANCEL);
[Do some one]
break;
case CONGESTION:
Noop(CONG);
[Do some one]
break;
case ANSWER:
Noop(ANS);
[Do some one]
break;
default:
Noop(default);
[Do some one]
break;
};
catch h {
Noop(Hangup in macro);
Noop(${DIALSTATUS});
Hangup;
};
return;
};
--
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
brya...@zktech.com wrote:
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 Harutyunyanhvarda...@gmail.com wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same
return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
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
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyanhvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
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 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
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyanhvarda...@gmail.com
*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
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 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, Nikhild.nik...@cem-solutions..net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM
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 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 mailto:d.nik...@cem-solutions.net wrote: 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 and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
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 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
Bryant Zimmerman wrote:
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
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 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
mailto:d.nik...@cem-solutions.net wrote:
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'
--
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Re: [asterisk-users] take input and store in variable
Carlos Chavez wrote:
On Mon, 2010-10-04 at 14:27 -0500, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab' user input, which was normally from ${EXTEN} but
I realize this won't work for extension 's'..
The short google search I did didn't turn up anything concrete.
What kind of search did you do to avoid getting the read command?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
--
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
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Re: [asterisk-users] 1.6 and 1.8 version A2Billing
Hello I have installed a2b 1.4 with Asterisk 1.6 - and working normal. Also have installed a2b 1.7.1 with Asterisk 1.6 - working fine (for me). -- 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 bilal ghayyad wrote: Hi All; Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4? Appreciate ur kindly help. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A2billing
Hello Change your a2b 1.8 version to a2b 1.7.1 version 1.7.1 version is a stable. And forum for a2b is http://forum.asterisk2billing.org/index.php -- 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 Flavio Miranda wrote: Hi, I am trying to configure a2billing 1.8 in my asterisk 1.6 but no value to DIALPREFIX and DESTINATION PREFIX is accepted when I try to create a RATE. thanks! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing
Hello You has installed a2b 1.7 version, and also had not do some permissions on folder and files. /usr/local/src/a2billing/admin/templates_c'. Be sure $compile_dir is writable by the web server user. in /usr/local/src/a2billing/common/lib/smarty/Smarty.class.php on line 1093 I think the best place to find some help and solution for a2b - its a2b forum. -- 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 César Pinto Magán wrote: Hello, You sould go to the admin page (a2billing/admin/). There are two possibles web pages for a2b: the admin page and the customer page. You should point to the one you like in each moment :) César Pinto Alhambra-Eidos *De:* asterisk-users-boun...@lists.digium.com en nombre de Flavio Miranda *Enviado el:* jue 16/09/2010 2:24 *Para:* Asterisk Asterisk *Asunto:* [asterisk-users] a2billing Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories: Index of /a2billing [ICO] Name http://200.220.138.116/a2billing/?C=N;O=DLast modified http://200.220.138.116/a2billing/?C=M;O=A Size http://200.220.138.116/a2billing/?C=S;O=A Description http://200.220.138.116/a2billing/?C=D;O=A [DIR] Parent Directory http://200.220.138.116/ - [DIR] admin/ http://200.220.138.116/a2billing/admin/ 15-Sep-2010 19:19 - [DIR] agent/ http://200.220.138.116/a2billing/agent/ 15-Sep-2010 19:21 - [DIR] common/ http://200.220.138.116/a2billing/common/ 15-Sep-2010 19:18 - [DIR] customer/ http://200.220.138.116/a2billing/customer/ 15-Sep-2010 19:20 - Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID disappear from CDR on transfer
Try to use local channel, and the pass the callerid of the caller to the local channel, an the later put this in CDR using h extention. -- 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 lechuck wrote: Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/ ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound trunk) 7) if john doe want to speak with caller assistant bridge the two lines using the transfer function of GXP2000 phone (REFER). After the transfer in the CDR i can't see the callerid of the caller, only data of the bridged call is reported. Any idea on what i can do to keep it ? thanks lechuck -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need information
Hello Why do not you want to use A2Billing with Asterisk? I think, for a start, it will give you everything to start a business. -- 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 mohamed daif wrote: Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
Hello I think you are even more complicated your job. In any case, in a2b v1.7 by default SIP peers in realtime and all agi-conf is in mysql (postgree). Please review manual and look a2b forums to get answers to Your questions. -- 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 Landy Landy wrote: I reinstalled a2billing, now 1.7. Created a trunk, call plan, rate card, added rate, and added rate to call plan. After creating a new customer (CC) now I was able to place a call through a2billing only for the new customers. In voip settings I added a SIP Config with the same information as in my current extensions since I would like to re-use these extension numbers to monitor them. Also changed the context for these to a2billing. When I try to call from these extension I get Enter your pin prompt. Now I'm stuck here. Other than inserting the record into the mysql table how can I espcify the account number and/or cc number and password for a new customer? Thanks. --- On Thu, 6/17/10, Vahan Yerkanianva...@arminco.com wrote: From: Vahan Yerkanianva...@arminco.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Thursday, June 17, 2010, 1:47 AM On 6/17/10 12:49 AM, Steve Edwards wrote: On Wed, 16 Jun 2010, Landy Landy wrote: I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php --SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. If you have CLI version of PHP installed, you can also try running /var/lib/asterisk/agi-bin/a2billing.php directly from the shell, and keep feeding it CR/LF, you'll see step-by-step variable assignment and hopefully the error message that stops it from working. You'll need display_errors on in php.ini for this as well. Most probably you're missing a PHP module or your SQL connection is failing. HTH, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
sip_iax_pstn_direct_call = NO ; enable the option to refill card with voucher in IVR (values : YES - NO) ivr_voucher = NO ; if ivr_voucher is active, you can define a prefix for the voucher number to refill your card ; values : number - don't forget to change prepaid-refill_card_with_voucher audio accordingly ivr_voucher_prefix = 8 ; When the user credit are below the minimum credit to call min_credit ; jump directly to the voucher IVR menu (values: YES - NO) jump_voucher_if_min_credit = NO ; Extracharge DIDs, multiple numbers and fees must be separated by comma ; extracharge_did = 1800XXX,1888XXX extracharge_did = ;extracharge_fee = 0.02,0.03 extracharge_fee = ; List the prefixes that will be stripped off if the call plan requires it international_prefixes = 9 ; More information about the Dial : http://voip-info.org/wiki-Asterisk+cmd+dial ; 30 : The timeout parameter is optional. If not specifed, the Dial command will wait indefinitely, exiting only when the originating channel hangs up, or all the dialed channels return a busy or error condition. Otherwise it specifies a maximum time, in seconds, that the Dial command is to wait for a channel to answer. ; H: Allow the caller to hang up by dialing * ; r: Generate a ringing tone for the calling party ; g: When the called party hangs up, exit to execute more commands in the current context. (new in 1.4) ; i: Asterisk will ignore any forwarding (302 Redirect) requests received. Essential for DID usage to prevent fraud. (new in 1.4) Useful if you are ringing a group of people and one person has set their phone to forwarded direct to voicemail on their cell or something which normally prevents any of the other phones from ringing. ; R: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. ; m: Provide Music on Hold to the calling party until the called channel answers. ; L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) ; %timeout% tag is replaced by the calculated timeout according the credit destination rate! ;dialcommand_param = |60|HRgrL(%timeout%:61000:3) ;dialcommand_param = |60|gL(%timeout%) dialcommand_param = |60|gS(%timeout%) ;dialcommand_param = |60|g ; by default (360 = 1HOUR MAX CALL) dialcommand_param_sipiax_friend = |60|HRgirL(360:61000:3) ; Define the order to make the outbound call ; YES - SIP/dialedphonenum...@gateway_ip - NO SIP/gateway_ip/dialedphonenumber ; Both should work exactly the same but i experimented one case when gateway was supporting dialedphonenum...@gateway_ip ; So in case of trouble, try it out switchdialcommand = yes ; failover recursive search - define how many time we want to authorize the research of the failover trunk when a call fails (value : 0 - 20) failover_recursive_limit = 2 ; For free calls, limit the duration: amount in seconds maxtime_tocall_negatif_free_route = 5400 ; Send a reminder email to the user when they are under min_credit_2call send_reminder = NO ; enable to monitor the call (to record all the conversations) ; value : YES - NO record_call = NO ; format of the recorded monitor file monitor_formatfile = gsm ; Force to play the balance to the caller in a predefined currency, to use the currency set for by the customer leave this field empty agi_force_currency = ; CURRENCY SECTION ; Define all the audio (without file extensions) that you want to play according to currency (use , to separate, ie usd:prepaid-dollar,mxn:pesos,eur:Euro,all:credit) currency_association = usd:dollars,mxn:pesos,eur:euros,all:credit ; Please enter the file name you want to play when we prompt the calling party to enter the destination number ; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011 file_conf_enter_destination = prepaid-enter-dest ; Please enter the file name you want to play when we prompt the calling party to choose the prefered language ; file_conf_enter_menulang = prepaid-menulang file_conf_enter_menulang = prepaid-menulang2 ; Define if you want to bill the 1st leg on callback even if the call is not connected to the destination callback_bill_1stleg_ifcall_notconnected = YES -- 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 Landy Landy wrote: Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On *Tue, 6/15/10, ram /talk2...@gmail.com/* wrote: From: ram talk2...@gmail.com Subject
Re: [asterisk-users] a2billing for residential voip usage
And also, what a2b version you are use? If you are use 1.7 then all config is in DB, if 1.3(4) all config in a2billing.conf -- 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 Vardan Harutyunyan wrote: I send you my a2b config for whole sale use_dnid = YES - this is the main option that you must use You can call this config like so: DeadAGI(a2billing.php|3) I hope this will be help you. [agi-conf3] ; the debug level ; 0=none, 1=low, 2=normal, 3=all debug = 0 ; Asterisk Version Information ; 1_1,1_2,1_4 By Default it will take 1_2 or higher asterisk_version = 1_4 ; Manage the answer on the call answer_call = NO ; Play audio - this will disable all stream file but not the Get Data ; for wholesale ensure that the authentication works and than number_try = 1 play_audio = NO ; play the goodbye message when the user has finished. say_goodbye = NO ; enable the menu to choose the language ; press 1 for English, pulsa 2 para el espaУБol, Pressez 3 pour FranУЇais play_menulanguage = NO ; force the use of a language, if you dont want to use it leave the option empty ; Values : ES, EN, FR, etc... (according to the audio you have installed) force_language = ; Introduction prompt : to specify an additional prompt to play at the beginning of the application intro_prompt = ; Minimum amount of credit to use the application min_credit_2call = 0 ; this is the minimum duration in seconds of a call in order to be billed ; any call with a length less than min_duration_2bill will have a 0 cost ; useful not to charge callers for system errors when a call was answered but it actually didn't connect min_duration_2bill = 0 ; if user doesn't have enough credit to call a destination, prompt him to enter another cardnumber notenoughcredit_cardnumber = NO ; if notenoughcredit_cardnumber = YES then assign the CallerID to the new cardnumber notenoughcredit_assign_newcardnumber_cid = NO ; if YES it will use the DNID and try to dial out, without asking for the phonenumber to call ; value : YES, NO use_dnid = YES ; list the dnid on which you want to avoid the use of the previous option use_dnid no_auth_dnid = 2400,2300 ; number of times the user can dial different number number_try = 1 ; this will force to select a specific call plan by the Rate Engine force_callplan_id = ; Play the balance to the user after the authentication (values : yes - no) say_balance_after_auth = NO ; Play the balance to the user after the call (values : yes - no) say_balance_after_call = NO ; Play the initial cost of the route (values : yes - no) say_rateinitial = NO ; Play the amount of time that the user can call (values : yes - no) say_timetocall = NO ; enable the setup of the callerID number before the outbound is made, by default the user callerID value will be use auto_setcallerid = NO ; If auto_setcallerid is enabled, the value of force_callerid will be set as CallerID force_callerid = ; If force_callerid is not set, then the following option ensures that CID is set to one of the card's configured caller IDs or blank if none available. ; NO - disable this feature, caller ID can be anything. ; CID - Caller ID must be one of the customers caller IDs ; DID - Caller ID must be one of the customers DID nos. ; BOTH - Caller ID must be one of the above two items. cid_sanitize = NO ; enable the callerid authentication ; if this option is active the CC system will check the CID of caller cid_enable = NO ; if the CID does not exist, then the caller will be prompt to enter his cardnumber cid_askpincode_ifnot_callerid = NO ; if the callerID authentication is enable and the authentication fails then the user will be prompt to enter his cardnumber ; this option will bound the cardnumber entered to the current callerID so that next call will be directly authenticate cid_auto_assign_card_to_cid = NO ; if the callerID is captured on a2billing, this option will create automatically a new card and add the callerID to it cid_auto_create_card = NO ; set the length of the card that will be auto create (ie, 10) cid_auto_create_card_len = 10 ; If cid_auto_create_card has been set to YES, the following options will define with which configuration we will create the card ; ; billing type of the new card ; ( value : POSTPAY or PREPAY) cid_auto_create_card_typepaid = POSTPAY ; amount of credit of the new card cid_auto_create_card_credit = 0 ; if postpay, define the credit limit for the card cid_auto_create_card_credit_limit = 1000 ; the tariffgroup to use for the new card (this is the ID that you can find on the admin web interface) cid_auto_create_card_tariffgroup = 6 ; to check callerID over the cardnumber authentication (to guard against spoofing
Re: [asterisk-users] a2billing for residential voip usage
look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- 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 Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re : Asterisk Call routing problem
Hangup is comming after Dial or AGI?
--
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
Adil Zaaraoui wrote:
Dear Vardan,
I had before the same problem, i reinstalled asterisk and it worked; now
i get the same problem;
i am using asterisk 1.4.22, it forwards if i call operator A (the one i
was testing); but all other numbers are not forwarded, it just hangup,
here is the output:
Accepting AUTHENTICATED call from 192.168.1.4:
requested format = unknown,
requested prefs = (ulaw|slin|alaw),
actual format = ulaw,
host prefs = (gsm|ulaw|alaw),
priority = mine
-- Executing [00212675410...@outbound:1] Set(IAX2/#000105-8303,
calleeNumber=011212675410113) in new stack
-- Executing [00212675410...@outbound:2] AGI(IAX2/#000105-8303)
HANGUP 'IAX2/#000105-8303'
thanks for the help
*De :* Vardan Harutyunyan hvarda...@gmail.com
*À :* asterisk-users@lists.digium.com
*Envoyé le :* Lun 14 juin 2010, 7h 36min 11s
*Objet :* Re: [asterisk-users] Asterisk Call routing problem
Whats about log?
--
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 mailto:i...@eif.am
www.eif-it.com
Adil Zaaraoui wrote:
Deal list,
I have a problem with asterisk call routing.
I configured my asterisk in a way that it forwards calls to a viop
provider using IAX2 protocol.
some cell phones numbers are routed and others are not!!! but when i try
directly zoiper with my provider it works perfectly.
Before, my asterisk was working perfectely. but i was testing just
calling one number, i think that asterisk keeps in its cache the
operator i was calling.
It seems like asterisk record in its cache the dialed numbers. i tried
to reload asterisk, i tried also to reboot the system, and the problem
still persist.
here is my config:
[outbound]
exten=_00X.,1,set(calleNumber=_011${EXTEN:2})
exten=_00X.,n,AGI(http://localhost/myScript.agi)
exten=_00X.,n,Dial(IAX2/mylo...@outbound/${calleNumber
mailto:IAX2/mylo...@outbound/${calleNumber})
exten=h,n,deadAGI(http://localhost/myScript2.agi)
i tried disabling my scripts and still getting the same problem.
Do you have any idea
--
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Re: [asterisk-users] Asterisk Call routing problem
Whats about log?
--
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
Adil Zaaraoui wrote:
Deal list,
I have a problem with asterisk call routing.
I configured my asterisk in a way that it forwards calls to a viop
provider using IAX2 protocol.
some cell phones numbers are routed and others are not!!! but when i try
directly zoiper with my provider it works perfectly.
Before, my asterisk was working perfectely. but i was testing just
calling one number, i think that asterisk keeps in its cache the
operator i was calling.
It seems like asterisk record in its cache the dialed numbers. i tried
to reload asterisk, i tried also to reboot the system, and the problem
still persist.
here is my config:
[outbound]
exten=_00X.,1,set(calleNumber=_011${EXTEN:2})
exten=_00X.,n,AGI(http://localhost/myScript.agi)
exten=_00X.,n,Dial(IAX2/mylo...@outbound/${calleNumber
mailto:IAX2/mylo...@outbound/${calleNumber})
exten=h,n,deadAGI(http://localhost/myScript2.agi)
i tried disabling my scripts and still getting the same problem.
Do you have any idea
--
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Re: [asterisk-users] CDR in case of CallForwarding
Hello
I have also became like this problems and have found solution to make
outgoing calls via local channel, and now if my customer do a transfer,
I can calculate extra international outgoing calls.
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
srinivas Antarvedi wrote:
Hello users,
i am looking for a solution in terms of CDR for the outbound only call.
presently i have the following setup.
//extensions.conf
[from-outside]
exten = _X.,1,NoOp(IncomingCall)
exten = _X.,n,BackGround(choce.wav)
exten = _X.,n,WaitExten(5)
exten = _X.,n,Hangup
exten = _1XX.,n,NoOp(1XX series Dialing)
exten = _1XX.,n,Dial(SIP/${EXTEN},60,rg)
exten = _1XX.,n,NoOp(${DIALSTATUS})
exten = _1XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CONGESTION | ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ]
?dialmobile:end)
exten = _1XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE
mobileNUMBER FOR THE us...@ougoingprovider,60,r)
exten = _1XX.,n(end),Hangup()
exten = _2XX.,n,NoOp(2XX series Dialing)
exten = _2XX.,n,Dial(SIP/${EXTEN},60,rg)
exten = _2XX.,n,NoOp(${DIALSTATUS})
exten = _2XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CONGESTION | ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ]
?dialmobile:end)
exten = _2XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE
mobileNUMBER FOR THE us...@ougoingprovider,60,r)
exten = _2XX.,n(end),Hangup()
//sip.conf
[outgoingprovider]
username=X
secret=y
port=
host=dfdfddf
fromuser=
- i am planning to take the number of calls made and the minutes spent
incase of mobile call forwarding
as it uses my outbound trunk by giving the accountcode set to a
particular call.
- but i am getting the total call (sip call + mobile call) as a single
record in my cdr record for a given accountcode.
- i need to get something like SIP/mobilenumber either in lastdata or
dstchannel associated accountcode as a separate cdr entry.
i tried with disabling cdr using NoCDR for the SIP call but for the
mobile call if i use ResetCDR() also i am totally
losing the callrecord.
- i tried with the ForkCDR() too but of no use..
is my requirement can be fulfilled by tweaking some changes in the
extensions.conf functions/applications??
please advise as i need to bill the user for the outbound calls only...
any help is sincerely appreciated. thanks in advance.
srinivas
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Re: [asterisk-users] Best way to limit outgoing calls per trunk
Hello, What version of Asterisk You are use? And what version of A2Billing You are use? If You use version 1.4.X of Asterisk You can put call-limit string in sip.conf for this trunk If You use A2B ver 1.7 and Asterk 1.4 you can announce this trunk using sip config in A2B, and the are call-limit via web. And how I know, in 1.6 is no more call-limit in sip.conf -- 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 bruce bruce wrote: Thanks for that. It very well detailed. I am not sure if I can use GROUP and GROUP_COUNT now that I see how it's used. You see, the call is placed by A2Billing so I don't have a control over setting GROUP increase and so if there is a call GROUP_COUNT won't work. I might resort back to using sed and awk to take output of core show channels and check for it's state. I will appreciate some guru of sed to to give me a true false for a channel up or not using sed and core show channels Thanks, Bruce On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman jonat...@thurmantech.com mailto:jonat...@thurmantech.com wrote: On Sun, May 30, 2010 at 9:37 AM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Thanks for the tip. I have been checking those two options. Would you be able to provide an example of how GROUP or GROUP_COUNT may check for a trunk usuage? Here is how I do it. It is based on Asterisk 1.6.1.x, and I created a generic sub-routine to call for limiting trunks to a specific number of calls. The code is documented, so it should give you a good idea of how to use it. http://thurmantech.com/node/7 -Jonathan From what I see is that you have to assing certain routes a group and then count the group, but how I do include a trunk in the group? Thanks On Sat, May 29, 2010 at 7:07 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Sat, 29 May 2010, bruce bruce wrote: I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: Using the GROUP/GROUP_COUNT functions in the dialplan is a better way. Using system() will mean creating a bunch of processes (each sed/awk/cut/etc command). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to limit outgoing calls per trunk
No, if You use call-limit the call will be dropped. How you put your customer on hold? If you use queue and the customer hear the music onhold, he will be billed for this connection I have try use queue and a2b, and I have do all connection using local channel, so I have become all is works, and the customer after speaking with agents and transferred to international number, is billed only for international call. Sorry for my english, if any question, please write. I will try to explain. Thanks -- 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 bruce bruce wrote: Thanks for the advice, but I have to keep the customer on hold till the line becomes available. Is that possible by the method you mentioned? I am using A2B 1.7 and Asterisk 1.4. Thanks, On Mon, May 31, 2010 at 2:27 AM, Vardan Harutyunyan hvarda...@gmail.com mailto:hvarda...@gmail.com wrote: Hello, What version of Asterisk You are use? And what version of A2Billing You are use? If You use version 1.4.X of Asterisk You can put call-limit string in sip.conf for this trunk If You use A2B ver 1.7 and Asterk 1.4 you can announce this trunk using sip config in A2B, and the are call-limit via web. And how I know, in 1.6 is no more call-limit in sip.conf -- 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 mailto:i...@eif.am www.eif-it.com http://www.eif-it.com bruce bruce wrote: Thanks for that. It very well detailed. I am not sure if I can use GROUP and GROUP_COUNT now that I see how it's used. You see, the call is placed by A2Billing so I don't have a control over setting GROUP increase and so if there is a call GROUP_COUNT won't work. I might resort back to using sed and awk to take output of core show channels and check for it's state. I will appreciate some guru of sed to to give me a true false for a channel up or not using sed and core show channels Thanks, Bruce On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman jonat...@thurmantech.com mailto:jonat...@thurmantech.com mailto:jonat...@thurmantech.com mailto:jonat...@thurmantech.com wrote: On Sun, May 30, 2010 at 9:37 AM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com mailto:bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Thanks for the tip. I have been checking those two options. Would you be able to provide an example of how GROUP or GROUP_COUNT may check for a trunk usuage? Here is how I do it. It is based on Asterisk 1.6.1.x, and I created a generic sub-routine to call for limiting trunks to a specific number of calls. The code is documented, so it should give you a good idea of how to use it. http://thurmantech.com/node/7 -Jonathan From what I see is that you have to assing certain routes a group and then count the group, but how I do include a trunk in the group? Thanks On Sat, May 29, 2010 at 7:07 PM, Steve Edwards asterisk.org http://asterisk.org http://asterisk.org@sedwards.com http://sedwards.com http://sedwards.com wrote: On Sat, 29 May 2010, bruce bruce wrote: I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: Using the GROUP/GROUP_COUNT functions in the dialplan is a better way. Using system() will mean creating a bunch of processes (each sed/awk/cut/etc command). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every
Re: [asterisk-users] Best way to limit outgoing calls per trunk
A ok, I think I have understand what you want.
The first, are you want that a2b calculate the buying price?
If it for you not so important, the you can use failover trunk in a2b.
Try this.
If no, then you can you dialplan to explain what he must do on hangup cause.
I use AEL. For example,
Dial(SIP/${AGENTSPHONE});
Noop(${DIALSTATUS});
switch(${DIALSTATUS}) {
case BUSY:
Noop( Busy);
Playback(${AGENT_ALLBUSY_MESSAGE});
break;
case CHANUNAVAIL:
Noop( Channel Unavailable);
Playback(${AGENT_UNAVAILABLE_MESSAGE});
break;
case NOANSWER:
Noop( No answer);
Playback(${AGENT_ALLBUSY_MESSAGE});
break;
case CANCEL:
Noop( Cancel);
Playback(${AGENT_ALLBUSY_MESSAGE});
break;
case CONGESTION:
Noop( Congestion);
Playback(${AGENT_UNAVAILABLE_MESSAGE});
break;
case ANSWER:
Noop( Answer);
break;
default:
Noop( Default);
Playback(${AGENT_UNAVAILABLE_MESSAGE});
break;
};
--
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
bruce bruce wrote:
Hi Vardan,
I am using use_dnid=yes and then setting the Account Code in Asterisk
dialplan before sending the call to A2Billing _x. context which
automatically dials. So, before the call goes to A2Billing, I can check
to see if there is a channel up or not. I am not sure how the local
channel you mentioned works. Would appreciate it if you share.
Can you determine the number of channels in the queue?
One of my trunks allows for 3 calls certain time of the day and sometime
it allows for only 1 channel. Hence the need for this.
Thanks,
On Mon, May 31, 2010 at 11:39 AM, Vardan Harutyunyan
hvarda...@gmail.com mailto:hvarda...@gmail.com wrote:
No, if You use call-limit the call will be dropped.
How you put your customer on hold?
If you use queue and the customer hear the music onhold, he will be
billed for this connection
I have try use queue and a2b, and I have do all connection using local
channel, so I have become all is works, and the customer after speaking
with agents and transferred to international number, is billed only for
international call.
Sorry for my english, if any question, please write. I will try to
explain.
Thanks
--
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 mailto:i...@eif.am
www.eif-it.com http://www.eif-it.com
bruce bruce wrote:
Thanks for the advice, but I have to keep the customer on hold
till the
line becomes available. Is that possible by the method you
mentioned? I
am using A2B 1.7 and Asterisk 1.4.
Thanks,
On Mon, May 31, 2010 at 2:27 AM, Vardan Harutyunyan
hvarda...@gmail.com mailto:hvarda...@gmail.com
mailto:hvarda...@gmail.com mailto:hvarda...@gmail.com wrote:
Hello,
What version of Asterisk You are use?
And what version of A2Billing You are use?
If You use version 1.4.X of Asterisk You can put call-limit
string in
sip.conf for this trunk
If You use A2B ver 1.7 and Asterk 1.4 you can announce this
trunk using
sip config in A2B, and the are call-limit via web.
And how I know, in 1.6 is no more call-limit in sip.conf
--
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 mailto:i...@eif.am mailto:i...@eif.am
mailto:i...@eif.am
www.eif-it.com http://www.eif-it.com http://www.eif-it.com
bruce bruce wrote:
Thanks for that. It very well detailed.
I am not sure if I can use GROUP and GROUP_COUNT now that I see
Re: [asterisk-users] CANCEL Reason
What you mean under this? You want do something after Dial by the CANCEL reason? -- 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 François BERGANZ wrote: Hello all, I need that Asterisk Always use Reason in a CANCEL. How to do? thank you *François * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About option U in Dial Ast version 1.6.2
OK I will try to explain my problems, sorry for my English. So, I want to indicate in dial plan, that the dialed channel is connectted. For example(extensions.conf): exten = s,1,Dial(ToSomeOne) exten = s,n,Hangup In this example, you can know about that the channel is connected after if hangup. So, if you do this: exten = s,1,Dial(ToSomeOne,,M(macros)) exten = s,n,Hangup [macro-macros] exten = s,1,Noop(Call conected) After Noop in macros, the dial plan is do nothing. This is a work. If I want make this using AEL (Asterisk 1.6.2), the macro is not working. While how we know, macro in AEL is changed to gosub. I can continue use the macros, that I have make in extensions.conf, but I dont want. So, in Dial application in asterisk 1.6.2 is now new option U. It the same, as M, but used gosub meanings. But he dont work same like M. Can any body help me to understand how this work. Thanks -- 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 Philipp von Klitzing wrote: Hi! as I understand, nobody not used this option? Would you like everyone on this list to give you an answer? ;- If you have a _real_ question about the U option of Dial(), then post it, and someone here might or might not have an answer. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
