asterisk users - Apr 2011 - Orginate not working well with PSTN lines

Dear all,

I am from Saudi  Arabiya and I am using digium hardware with asterisk 1.6.

When I am executing following AMI originate API. Orginate start to
execute extenstion without knowing of PSTN(FXO) channel is ringing.

Any one can help me to  resolve this issue ?

Action: Originate
Channel: Dahdi/g0/2923878
Context: outbound-ivr
Exten: 1234
Priority: 1
ActionID: ABC45678901234567890


Response: Success
ActionID: ABC45678901234567890
Message: Originate successfully queued


  -- Remote UNIX connection disconnected
       > Channel DAHDI/1-1 was answered.
    -- Executing [1234 at outbound-ivr:1] SayDigits("DAHDI/1-1",
"1234")
in new stack
    -- <DAHDI/1-1> Playing 'digits/1.gsm' (language 'en')
    -- <DAHDI/1-1> Playing 'digits/2.gsm' (language 'en')
    -- <DAHDI/1-1> Playing 'digits/3.gsm' (language 'en')
    -- <DAHDI/1-1> Playing 'digits/4.gsm' (language 'en')
    -- Executing [1234 at outbound-ivr:2] Playback("DAHDI/1-1",
"demo-congrats") in new stack
    -- <DAHDI/1-1> Playing 'demo-congrats.gsm' (language
'en')
    -- Executing [1234 at outbound-ivr:3] Hangup("DAHDI/1-1",
"") in new stack
  == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on
'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'


Thanks & Regards,
Ashik

"Originate successfully queued" only means that the originate action
was handed off to asterisk not that is was executed yet. There are other events,
depending on which events you are "reading", that tell you the call
was answered and such.
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/



On Apr 26, 2011, at 2:43 AM, Ashik Ali wrote:
> Dear all,
> 
> I am from Saudi  Arabiya and I am using digium hardware with asterisk 1.6.
> 
> When I am executing following AMI originate API. Orginate start to
> execute extenstion without knowing of PSTN(FXO) channel is ringing.
> 
> Any one can help me to  resolve this issue ?
> 
> Action: Originate
> Channel: Dahdi/g0/2923878
> Context: outbound-ivr
> Exten: 1234
> Priority: 1
> ActionID: ABC45678901234567890
> 
> 
> Response: Success
> ActionID: ABC45678901234567890
> Message: Originate successfully queued
> 
> 
>  -- Remote UNIX connection disconnected
>> Channel DAHDI/1-1 was answered.
>    -- Executing [1234 at outbound-ivr:1] SayDigits("DAHDI/1-1",
"1234")
> in new stack
>    -- <DAHDI/1-1> Playing 'digits/1.gsm' (language
'en')
>    -- <DAHDI/1-1> Playing 'digits/2.gsm' (language
'en')
>    -- <DAHDI/1-1> Playing 'digits/3.gsm' (language
'en')
>    -- <DAHDI/1-1> Playing 'digits/4.gsm' (language
'en')
>    -- Executing [1234 at outbound-ivr:2] Playback("DAHDI/1-1",
> "demo-congrats") in new stack
>    -- <DAHDI/1-1> Playing 'demo-congrats.gsm' (language
'en')
>    -- Executing [1234 at outbound-ivr:3] Hangup("DAHDI/1-1",
"") in new stack
>  == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on
'DAHDI/1-1'
>    -- Hungup 'DAHDI/1-1'
> 
> 
> Thanks & Regards,
> Ashik
> 
> --
> _____________________________________________________________________
> -- 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
> 
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Hello.

Considering the following setup: 

Legacy PBX --(ISDN)--> Asterisk --(MFC/R2)--> PSTN


When a user dials out, Asterisk receive overlap digits, matches them to an
extension and dial the PSTN, completing the call. So far so good.

The issue I'm trying to solve (or at least improve) is the call takes much
longer to complete than the users were used to before having Asterisk between
the PBX and the PSTN. It happens because the digits are sent to the PSTN only
after the extension is matched in the dialplan, and dialing on MFC/R2 takes a
few seconds.

Here's the console log. Notice how it takes 6 seconds from the instant the
user starts dialing to the instant the dialed number starts to ring. First 3
seconds is the user manually dialing plus Asterisk absolute timeout. Next 3
seconds are the time Asterisk takes to dial the number to the PSTN and the call
be accepted.


[Apr 26 10:57:13]     -- Accepting overlap call from '7416' to
'<unspecified>' on channel 0/1, span 2
[Apr 26 10:57:13]     -- Starting simple switch on 'DAHDI/32-1'
            
             *** User finished dialing + Asterisk absolute timeout ***

[Apr 26 10:57:16]     -- Executing [0145333114657 at pbx:1]
Answer("DAHDI/33-1", "") in new stack
[Apr 26 10:57:16]     -- Executing [0145333114657 at pbx:2]
Dial("DAHDI/33-1", "DAHDI/g1/0145333114657") in new stack
[Apr 26 10:57:16]     -- Called g1/0145333114657

             *** Asterisk starts dialing to the PSTN ***

[Apr 26 10:57:19] MFC/R2 call has been accepted on forward channel 1
[Apr 26 10:57:19]     -- DAHDI/1-1 is ringing

             *** Dialed number finally rings ***

So my question is: is there a way to fully overlap the digits from the
user's phone on the PBX (ISDN) to the PSTN (MFC/R2), eliminating the need to
wait for an extension to be matched? I already have overlapdial=yes in both
spans, but that didn't made it. Also googled for it, even searched this list
archives but found nothing.


chan_dahdi.conf:

[channels]

signalling=mfcr2
mfcr2_variant=br
mfcr2_get_ani_first=no
mfcr2_max_ani=20
mfcr2_max_dnis=4
mfcr2_category=national_subscriber
mfcr2_logdir=span1
mfcr2_call_files=no
mfcr2_logging=all
mfcr2_mfback_timeout=-1
mfcr2_metering_pulse_timeout=-1
mfcr2_allow_collect_calls=yes
mfcr2_double_answer=no
mfcr2_immediate_accept=no
mfcr2_forced_release=no
mfcr2_charge_calls=yes
language=pt_BR
echocancel=yes
echocancelwhenbridged=no
callgroup=0
pickupgroup=0
group=1
context=telco
overlapdial=yes
channel => 1-15,17-31

switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
priindication=outofband
signalling=pri_net
busydetect=yes
busycount=5
language=pt_BR
echocancel=yes
echocancelwhenbridged=no
overlapdial=yes
group=2
context=pbx
channel => 32-46,48-62



extensions.conf:

[telco]
exten => _X.,1,Dial(DAHDI/g2/${EXTEN})

[pbx]
exten => _X.,1,Dial(DAHDI/g1/{$EXTEN})

Thanks for your solution.

Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco  ? .

Does anybody successed; to make asterisk to detect ring back tone from
PSTN telco ?

Thanks,
Ashik

On Wed, Apr 27, 2011 at 12:44 PM, Gilles <codecomplete at free.fr>
wrote:
> On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali
> <beaasteriskguru at gmail.com> wrote:
>>The problem here is that as soon as asterisk dialing on fxo lines it
>>sets channel status as "answered" ?although the chennel is
getting
>>ring back tone from
>>other party.
>>
>>Anyone can suggest me to solve this issue ?
>
> The only solution I know is to have Asterisk play a message in a loop
> for eg. 1mn, prompting the callee to hit a key to let the server know
> that the call was 1) answered 2) by a human being.
>
>
> --
> _____________________________________________________________________
> -- 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
>

When dialing is finished on an analog FXO Asterisk considers it answered.  The
solution is to use something that is not an analog FXO like PRI or SIP to a
carrier.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Ashik Ali
Sent: Wednesday, April 27, 2011 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Orginate not working well with PSTN lines

Thanks for your solution.

Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco  ? .

Does anybody successed; to make asterisk to detect ring back tone from
PSTN telco ?

Thanks,
Ashik

On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
<beaasteriskguru at gmail.com> wrote:
>Anybody can explain me why asterisk is unable to detect ringback tone
>from PSTN telco  ? .
I guess it was a lot of work, and nobody bothered adding this to the
Zaptel driver.

I thank everyone, for their fruitfull informations.

Regards,
Ashik Ali

On Fri, Apr 29, 2011 at 2:04 AM, Gilles <codecomplete at free.fr>
wrote:
> On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
> <beaasteriskguru at gmail.com> wrote:
>>Anybody can explain me why asterisk is unable to detect ringback tone
>>from PSTN telco ?? .
>
> I guess it was a lot of work, and nobody bothered adding this to the
> Zaptel driver.
>
>
> --
> _____________________________________________________________________
> -- 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
>

On Fri, Apr 29, 2011 at 01:04:42AM +0200, Gilles wrote:
> On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
> <beaasteriskguru at gmail.com> wrote:
>>
>> Anybody can explain me why asterisk is unable to detect ringback tone
from
>> PSTN telco  ? .
> 
> I guess it was a lot of work, and nobody bothered adding this to the
> Zaptel driver.
I know this thread is dead but: I do not believe this should go into the DAHDI
kernel modules.

The only thing the kernel modules could possibly do if the "ring" tone
is
detected is queue an event on the channel for Asterisk to decide how to
handle. Asterisk / chan_dahdi is already typically monitoring the channel for
DTMF digits and looking for additional tones and patterns could be added
there.  Asterisk would potentially need to have the tonezones of all possible
destinations loaded which would make this complex and resource hungry.

Cheers,
Shaun

--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org