[asterisk-users] 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/${EXTEN},40,t)
same=> n,Hangup
exten => i,1,Playback(invalid)
same=> n,Goto(IVRINMA,s,2)
exten=> t,1,Dial(SIP/101,38,t)
exten=> t,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?2,1:)
exten => 2,1,Dial(SIP/102,38,t)
same=> n,Hangup()
## the second option, if possible ###
I have a gw wiht 4 port gsm , my provider gives me 4 lines and one of
them is the main , the problem is that all my incoming calls using
this number and is always busy , and the other three are always free,
it is possible that the call is transferred to another channel?
Channel 1 : XXX1 "Main Number"
Channel 2 : XXX2 "other"
Channel 3 : XXX3 "other"
Channel 4 : XXX4 "other"
regardss
--
rickygm
http://gnuforever.homelinux.com
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Re: [asterisk-users] situation with ivr and four-channel gateway
I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not
'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten =>
101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40))
same => n,Dial(SIP/102,40,t)
same => n,Hangup()
On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez
wrote:
> Hi list, I need your help ,I have an incoming call x the ivr and the
> operator takes the call. ext "101" , If a second call reenters and the
> operator is talking, I want to send to the extension 102 I use the
> Variable DIALSTATUS , but not working
>
> check IVR
>
> [IVRINMA]
>
> exten => s,1,Wait(1)
> exten => s,n,Set(CHANNEL(language)=es)
> same=> n,Set(TIMEOUT(digit)=4)
> same=> n,Set(TIMEOUT(response)=5)
> same=> n,Wait(1)
> same=> n,Background(/tmp/ivr/menu)
> same=> n,WaitExten(5)
> exten => 0,1,Playback(pls-wait-connect-call)
> exten => 0,n,Goto(operadora,101,1)
> exten => _10[1-3],1,Dial(SIP/${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 : XXX1 "Main Number"
> Channel 2 : XXX2 "other"
> Channel 3 : XXX3 "other"
> Channel 4 : XXX4 "other"
>
> regardss
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] situation with ivr and four-channel gateway
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 sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-25 18:23 GMT-06:00 John Kiniston :
> 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
--
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Re: [asterisk-users] situation with ivr and four-channel gateway
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 : XXX1 "Main Number"
> Channel 2 : XXX2 "other"
> Channel 3 : XXX3 "other"
> Channel 4 : XXX4 "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 .
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Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-26 10:45 GMT-06:00 A J Stiles :
>
> 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
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Re: [asterisk-users] situation with ivr and four-channel gateway
On Thursday 26 Feb 2015, ricky gutierrez wrote: > 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
Re: [asterisk-users] situation with ivr and four-channel gateway
2015-02-27 10:25 GMT-06:00 A J Stiles :
> 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 => _NXXX,n,Dial(SIP/1003/${EXTEN},55,rT)
exten => _NXXX,n,Dial(SIP/1004/${EXTEN},55,rT)
exten => _NXXX,n,Dial(SIP/1001/${EXTEN},55,rT)
exten => _NXXX,n,Dial(SIP/1002/${EXTEN},55,rT)
exten => _NXXX,n,Playback(all-circuits-busy-now)
exten => _NXXX,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.
--
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http://gnuforever.homelinux.com
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Re: [asterisk-users] situation with ivr and four-channel gateway
On Friday 27 Feb 2015, ricky gutierrez wrote: > 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: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] situation with ivr and four-channel gateway
2015-03-02 3:44 GMT-06:00 A J Stiles :
> 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
> 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
> 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.
--
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