asterisk users - Feb 2015 - situation with ivr and four-channel gateway

Hi list, I need your help ,I have an incoming call x the ivr and the
operator takes the call. ext "101" , If a second call reenters and the
operator is talking, I want to send to the extension 102 I use the
Variable DIALSTATUS , but not working

check IVR

[IVRINMA]

exten => s,1,Wait(1)
exten => s,n,Set(CHANNEL(language)=es)
same=> n,Set(TIMEOUT(digit)=4)
same=> n,Set(TIMEOUT(response)=5)
same=> n,Wait(1)
same=> n,Background(/tmp/ivr/menu)
same=> n,WaitExten(5)
exten => 0,1,Playback(pls-wait-connect-call)
exten => 0,n,Goto(operadora,101,1)
exten => _10[1-3],1,Dial(SIP/${EXTEN},40,t)
same=> n,Hangup
exten => i,1,Playback(invalid)
same=> n,Goto(IVRINMA,s,2)
exten=> t,1,Dial(SIP/101,38,t)
exten=> t,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?2,1:)
exten => 2,1,Dial(SIP/102,38,t)
same=> n,Hangup()

## the second option, if possible ###

I have a gw  wiht 4 port gsm , my provider gives me 4 lines and one of
them is the main , the problem is that all my incoming calls using
this number and is always busy , and the other three are always free,
it is possible that the call is transferred to another channel?

Channel 1 : XXXXXXX1 "Main Number"
Channel 2 : XXXXXXX2 "other"
Channel 3 : XXXXXXX3 "other"
Channel 4 : XXXXXXX4 "other"

regardss

-- 
rickygm

http://gnuforever.homelinux.com

I'd recommend using DEVICE_STATE

On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not
'NOT_INUSE' then dial it, Otherwise dial SIP/102

exten =>
101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40))
 same =>       n,Dial(SIP/102,40,t)
 same =>       n,Hangup()

On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez <xserverlinux at
gmail.com>
wrote:
> Hi list, I need your help ,I have an incoming call x the ivr and the
> operator takes the call. ext "101" , If a second call reenters
and the
> operator is talking, I want to send to the extension 102 I use the
> Variable DIALSTATUS , but not working
>
> check IVR
>
> [IVRINMA]
>
> exten => s,1,Wait(1)
> exten => s,n,Set(CHANNEL(language)=es)
> same=> n,Set(TIMEOUT(digit)=4)
> same=> n,Set(TIMEOUT(response)=5)
> same=> n,Wait(1)
> same=> n,Background(/tmp/ivr/menu)
> same=> n,WaitExten(5)
> exten => 0,1,Playback(pls-wait-connect-call)
> exten => 0,n,Goto(operadora,101,1)
> exten => _10[1-3],1,Dial(SIP/${EXTEN},40,t)
> same=> n,Hangup
> exten => i,1,Playback(invalid)
> same=> n,Goto(IVRINMA,s,2)
> exten=> t,1,Dial(SIP/101,38,t)
> exten=> t,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?2,1:)
> exten => 2,1,Dial(SIP/102,38,t)
> same=> n,Hangup()
>
> ## the second option, if possible ###
>
> I have a gw  wiht 4 port gsm , my provider gives me 4 lines and one of
> them is the main , the problem is that all my incoming calls using
> this number and is always busy , and the other three are always free,
> it is possible that the call is transferred to another channel?
>
> Channel 1 : XXXXXXX1 "Main Number"
> Channel 2 : XXXXXXX2 "other"
> Channel 3 : XXXXXXX3 "other"
> Channel 4 : XXXXXXX4 "other"
>
> regardss
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
>
> --
> _____________________________________________________________________
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On Wed, 25 Feb 2015, John Kiniston wrote:
> I'd recommend using DEVICE_STATE
> 
> On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's 
> not 'NOT_INUSE' then dial it, Otherwise dial SIP/102
> 
> exten =>
101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40))
> ?same =>?????? n,Dial(SIP/102,40,t)
> ?same =>?????? n,Hangup()
Remember to set 'callcounter = yes' in sip.conf.

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

2015-02-25 18:23 GMT-06:00 John Kiniston <johnkiniston at
gmail.com>:
> I'd recommend using DEVICE_STATE
>
> On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's
not
> 'NOT_INUSE' then dial it, Otherwise dial SIP/102
>
> exten =>
>
101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40))
>  same =>       n,Dial(SIP/102,40,t)
>  same =>       n,Hangup()
>

Hi john and Steve , I do tests with advice

-- 
rickygm

http://gnuforever.homelinux.com

On Wednesday 25 Feb 2015, ricky gutierrez wrote:
> I have a gw  wiht 4 port gsm , my provider gives me 4 lines and one of
> them is the main , the problem is that all my incoming calls using
> this number and is always busy , and the other three are always free,
> it is possible that the call is transferred to another channel?
> 
> Channel 1 : XXXXXXX1 "Main Number"
> Channel 2 : XXXXXXX2 "other"
> Channel 3 : XXXXXXX3 "other"
> Channel 4 : XXXXXXX4 "other"
You just need to use call groups.

In your chan_extra.conf  (if it's an OpenVox)  or chan_dahdi.conf, add 
something like
  group=1
to the definition for each span.

Now in the  [globals]  section of your dialplah, have something like
  MOBILE=EXTRA/r1
for an OpenVox card, or
  MOBILE=DAHDI/r1
for other makes.  Now you need your Dial() statements to be something like
  Dial(${MOBILE}/${EXTEN},180

Calls will then be made by trying each span in turn until an available one is 
found.  So if you have an incoming call on span 1, Asterisk will try spans 2, 
3 and 4 in turn before giving up.  It also will remember which span it used 
last, and start with the next one next time; so the calls should be 
distributed roughly evenly across your SIMs.

For more information about this  (and some other modes you can use which do 
slightly different things than "r"),  see
  http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
(yes, it refers to Zaptel; but the syntax is the same for DAHDI and EXTRA 
channels).

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

2015-02-26 10:45 GMT-06:00 A J Stiles <asterisk_list at
earthshod.co.uk>:
>
> You just need to use call groups.
>
> In your chan_extra.conf  (if it's an OpenVox)  or chan_dahdi.conf, add
> something like
>   group=1
> to the definition for each span.
>
> Now in the  [globals]  section of your dialplah, have something like
>   MOBILE=EXTRA/r1
> for an OpenVox card, or
>   MOBILE=DAHDI/r1
> for other makes.  Now you need your Dial() statements to be something like
>   Dial(${MOBILE}/${EXTEN},180
>
> Calls will then be made by trying each span in turn until an available one
is
> found.  So if you have an incoming call on span 1, Asterisk will try spans
2,
> 3 and 4 in turn before giving up.  It also will remember which span it used
> last, and start with the next one next time; so the calls should be
> distributed roughly evenly across your SIMs.
>
> For more information about this  (and some other modes you can use which do
> slightly different things than "r"),  see
>   http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
> (yes, it refers to Zaptel; but the syntax is the same for DAHDI and EXTRA
> channels).
>
Hi A J , I have a sangoma gsm gateway "4"channels  , not use chan
dahdi




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