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xserverlinux at gmail.com
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 Posted: Wed Feb 25, 2015 4:08 pm Post subject: [asterisk-users] situation with ivr and four-channel gateway |
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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/${EXTEN},40,t)
same=> n,Hangup
exten => i,1,Playback(invalid)
same=> n,Goto(IVRINMA,s,2)
exten=> t,1,Dial(SIP/101,38,t)
exten=> t,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?2,1
exten => 2,1,Dial(SIP/102,38,t)
same=> n,Hangup()
## the second option, if possible ###
I have a gw wiht 4 port gsm , my provider gives me 4 lines and one of
them is the main , the problem is that all my incoming calls using
this number and is always busy , and the other three are always free,
it is possible that the call is transferred to another channel?
Channel 1 : XXXXXXX1 "Main Number"
Channel 2 : XXXXXXX2 "other"
Channel 3 : XXXXXXX3 "other"
Channel 4 : XXXXXXX4 "other"
regardss
--
rickygm
http://gnuforever.homelinux.com
--
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johnkiniston at gmail.com
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| Posted: Wed Feb 25, 2015 7:23 pm Post subject: [asterisk-users] situation with ivr and four-channel gateway |
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I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not 'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten => 101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40))
same => n,Dial(SIP/102,40,t)
same => n,Hangup()
On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez <xserverlinux@gmail.com (xserverlinux@gmail.com)> wrote:
| Quote: | Hi list, I need your help ,I have an incoming call x the ivr and the
operator takes the call. ext "101" , If a second call reenters and the
operator is talking, I want to send to the extension 102 I use the
Variable DIALSTATUS , but not working
check IVR
[IVRINMA]
exten => s,1,Wait(1)
exten => s,n,Set(CHANNEL(language)=es)
same=> n,Set(TIMEOUT(digit)=4)
same=> n,Set(TIMEOUT(response)=5)
same=> n,Wait(1)
same=> n,Background(/tmp/ivr/menu)
same=> n,WaitExten(5)
exten => 0,1,Playback(pls-wait-connect-call)
exten => 0,n,Goto(operadora,101,1)
exten => _10[1-3],1,Dial(SIP/${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
--
_____________________________________________________________________
-- 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:
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--
A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects.
---Heinlein |
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asterisk.org at sedwar...
Guest
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| Posted: Wed Feb 25, 2015 10:02 pm Post subject: [asterisk-users] situation with ivr and four-channel gateway |
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On Wed, 25 Feb 2015, John Kiniston wrote:
| Quote: | 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@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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xserverlinux at gmail.com
Guest
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| Posted: Thu Feb 26, 2015 10:55 am Post subject: [asterisk-users] situation with ivr and four-channel gateway |
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2015-02-25 18:23 GMT-06:00 John Kiniston <johnkiniston@gmail.com>:
| Quote: | 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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk_list at earth...
Guest
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| Posted: Thu Feb 26, 2015 11:45 am Post subject: [asterisk-users] situation with ivr and four-channel gateway |
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On Wednesday 25 Feb 2015, ricky gutierrez wrote:
| Quote: | 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 .
--
_____________________________________________________________________
-- 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
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xserverlinux at gmail.com
Guest
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| Posted: Thu Feb 26, 2015 5:52 pm Post subject: [asterisk-users] situation with ivr and four-channel gateway |
|
|
2015-02-26 10:45 GMT-06:00 A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: |
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
http://gnuforever.homelinux.com
--
_____________________________________________________________________
-- 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
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asterisk_list at earth...
Guest
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| Posted: Fri Feb 27, 2015 11:26 am Post subject: [asterisk-users] situation with ivr and four-channel gateway |
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On Thursday 26 Feb 2015, ricky gutierrez wrote:
| Quote: | Hi A J , I have a sangoma gsm gateway "4"channels , not use chan dahdi
|
O.K. So what does your existing Dial() statement in extensions.conf look
like?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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 |
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xserverlinux at gmail.com
Guest
|
| Posted: Fri Feb 27, 2015 6:01 pm Post subject: [asterisk-users] situation with ivr and four-channel gateway |
|
|
2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | O.K. So what does your existing Dial() statement in extensions.conf look
like?
|
apology, put the gateway was sangoma but is a openvox ,
all my outgoing calls out for this context:
[my-mobile-out]
exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT)
exten => _NXXXXXXX,n,Dial(SIP/1004/${EXTEN},55,rT)
exten => _NXXXXXXX,n,Dial(SIP/1001/${EXTEN},55,rT)
exten => _NXXXXXXX,n,Dial(SIP/1002/${EXTEN},55,rT)
exten => _NXXXXXXX,n,Playback(all-circuits-busy-now)
exten => _NXXXXXXX,n,Hangup()
my main number is registered on "1002" channel gsm 1
the problem is that my pbx all incoming calls using only the channel
gsm 1 , the idea is that an incoming call to channel 1 is passed to
channel 2
regardss.
--
rickygm
http://gnuforever.homelinux.com
--
_____________________________________________________________________
-- 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:
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asterisk_list at earth...
Guest
|
| Posted: Mon Mar 02, 2015 4:45 am Post subject: [asterisk-users] situation with ivr and four-channel gateway |
|
|
On Friday 27 Feb 2015, ricky gutierrez wrote:
| Quote: | the problem is that my pbx all incoming calls using only the channel
gsm 1 , the idea is that an incoming call to channel 1 is passed to
channel 2
|
Ah. *Incoming* calls are not something that is within your control; they have
already been routed onto a line by your telco. So you will need to speak to
someone at your telco about doing this.
As a temporary measure, you could try setting up divert-on-busy so SIM1
diverts to SIM2, SIM2 diverts to SIM3, SIM3 diverts to SIM4 and SIM4 diverts
to SIM1. You can do this with specially-crafted Dial() statements, or by
temporarily inserting the SIMs in an old mobile phone. See your telco's
website for details of setting up call diversion.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
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xserverlinux at gmail.com
Guest
|
| Posted: Mon Mar 02, 2015 11:30 pm Post subject: [asterisk-users] situation with ivr and four-channel gateway |
|
|
2015-03-02 3:44 GMT-06:00 A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | Ah. *Incoming* calls are not something that is within your control; they have
already been routed onto a line by your telco. So you will need to speak to
someone at your telco about doing this.
|
Hi Aj, I call to telco and say they can not in GSM, only on lines are analogous
| Quote: | As a temporary measure, you could try setting up divert-on-busy so SIM1
diverts to SIM2, SIM2 diverts to SIM3, SIM3 diverts to SIM4 and SIM4 diverts
to SIM1. You can do this with specially-crafted Dial() statements,
|
With asterisk or the openvox gw?
or by
| Quote: | temporarily inserting the SIMs in an old mobile phone. See your telco's
website for details of setting up call diversion.
|
these guys do not help much! .
the ivr worked perfect with DEVICE_STATE , thank john!
exten => t,1,ExecIf($[ ${DEVICE_STATE(SIP/${EXTEN})} = INUSE ]?Busy)
exten => t,n,Dial(SIP/110,38,t)
same=> n,Dial(SIP/162,40,t)
same=> n,Hangup()
thnk for all help.
--
rickygm
http://gnuforever.homelinux.com
--
_____________________________________________________________________
-- 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:
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