[asterisk-users] dtmf Caller-id detection before first ring
Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want to dig it and make it work in asterisk. 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf Caller-id detection before first ring
We don't have polarity reversal before first ring I think so. Not confirmed.
I able to see and hear dtmf and ring tones while playing recorded
wave file using audacity. As per the instruction given by Mr. Pezhman
Lai. I found somthing
while googling . The bug instructions as follows
https://issues.asterisk.org/view.php?id=9096&nbn=2.
I am using 8 port digium tdm card. So I decided edit
wctdm24xxp/base.c as per the instruction written in above issue. As
of now I didn't touch asterisk chan_dahdi.c.
I modified base.c code as follows,
static void wctdm_dtmfcheck_fakepolarity(struct wctdm *wc, int card, u8 sample8)
{
u32 sample16;
struct fxo *const fxo = &(wc->mods[card].fxo);
/* only look for sound on the line if dtmf flag is on, it is an fxo
* card and line is onhook */
if (!dtmf || !(wc->cardflag & (1 << card)) ||
(wc->modtype[card] != MOD_TYPE_FXO) || fxo->offhook) {
return;
}
/* don't look for noise if we're already processing it, or there is a
* ringing tone */
if (!fxo->readcid && !fxo->wasringing &&
wc->intcount > fxo->cidtimer + 400) {
sample16 = DAHDI_XLAW(sample8, wc->chans[card]);
if (sample16 > 2000 || sample16 < -2000) {
fxo->readcid = 1;
fxo->cidtimer = wc->intcount;
if (debug && ( card == 2 )) {
printk(KERN_DEBUG "DTMF CLIP on
<<<%i>>> <%X>\n",
card + 1,sample16);
}
// dahdi_qevent_lock(wc->chans[card],
//DAHDI_EVENT_POLARITY);
}
} else if (fxo->readcid && wc->intcount > fxo->cidtimer + 2000) {
/* reset flags if it's been a while */
fxo->cidtimer = wc->intcount;
fxo->readcid = 0;
}
}
After compilation of above, I just restarted dahdi and monitored
kernel message. I got following messages before receiving call
DTMF CLIP on <<<3>>> <68>
DTMF CLIP on <<<3>>> <84>
DTMF CLIP on <<<3>>> <58>
DTMF CLIP on <<<3>>> <78>
DTMF CLIP on <<<3>>> <60>
DTMF CLIP on <<<3>>> <48>
DTMF CLIP on <<<3>>> <60>
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on <<<3>>> <84>
DTMF CLIP on <<<3>>> <78>
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on <<<3>>> <68>
DTMF CLIP on <<<3>>> <68>
DTMF CLIP on <<<3>>> <70>
DTMF CLIP on <<<3>>>
DTMF CLIP on <<<3>>> <70>
But some calls, I am getting long hexadecimall value as follows,
DTMF CLIP on <<<3>>> <48>
DTMF CLIP on <<<3>>>
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on <<<3>>> <84>
DTMF CLIP on <<<3>>> <78>
RING on 1/3!
NO RING on 1/3!
Can u guide me on right Mr.Lali and Cohen.. ?
Thanks & Regards,
Ashik Ali
On Sun, May 29, 2011 at 12:13 PM, Tzafrir Cohen
wrote:
> On Sat, May 28, 2011 at 02:34:36PM +0300, Ashik Ali wrote:
>> Hi dears,
>>
>> I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and
>> Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) .
>>
>> I am facing problem with detecting caller id before first ring.I
>> recorded the dahdi channel using dahdi_monitor command. Where I am
>> able to see and hear caller-id dtmf tones.
>
> Is there a polarity reversal before the caller ID string is sent?
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> 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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Re: [asterisk-users] Asterisk Integration with Android device
Hi, Have u tried NAT(Network Address Translation Settings) ?. place the following setting on SIP account. nat=yes externip=nnn.nnn.nnn.nnn ; externhost= localnet=192.168.1.0/255.255.255.0 externrefresh=10 for more details hv look on the http://www.wirelessforums.org/uk-telecom-voip/problem-getting-asterisk-behind-nat-run-sipproxd-56038.html Hope it helps !. Thanks, Ashik Ali On Thu, Aug 25, 2011 at 8:19 PM, amit anand wrote: > Hi > find the answer inline > > On Thu, Aug 25, 2011 at 21:24, Gopal krishnan > wrote: >> >> Thanks for all your comments. Actually I have 3G connection but even then >> the signal in my mobile automatically changes from 3g to 2G; it is >> automatically going to Edge signal. Anyways let me try with some other >> softphone like media5. >> Regards, >> Gopal >> >> On Thu, Aug 25, 2011 at 9:21 AM, James Perkins wrote: >>> >>> Try media5 fone. >>> I couldn't get 3cx to work on my iphone and tried about 7 different >>> softfones. Media5 is the best by a long shot. >>> Android is still in better and haven't tried it but if its anything like >>> their iphone app it will be worth a look. >>> There is a signup for the better at the website. >>> let us know how you go. >>> James >>> >>> - Original Message - >>> From: bakko >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Sent: Wednesday, August 24, 2011 11:48 PM >>> Subject: Re: [asterisk-users] Asterisk Integration with Android device >>> I think don't work with 2G network. >>> >>> Regards >>> >>> - Original Message - >>> From: Gopal krishnan >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Sent: Wednesday, August 24, 2011 4:01 PM >>> Subject: [asterisk-users] Asterisk Integration with Android device >>> Hi, >>> I created a extension in Asterisk, the extension has been configured in >>> Android softphone 3cx. When I tried to call from Andorid phone to some other >>> IP extension which is registered in Asterisk, I am not able to hear the >>> voice, when I check the asterisk log or wireshark there is only one way RTP >>> traffic, from Android I am connecting to Asterisk via 2G GSM network. >>> Any idea would be appreciated. >>> Regards, >>> Gopal >>> >>> >>> >>> -- >>> _ >>> -- 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 > > > Try sipdroid for android > -- > > Amit Anand > > +91 9818559898 > > > -- > _ > -- 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
[asterisk-users] 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@outbound-ivr:1] SayDigits("DAHDI/1-1", "1234")
in new stack
-- Playing 'digits/1.gsm' (language 'en')
-- Playing 'digits/2.gsm' (language 'en')
-- Playing 'digits/3.gsm' (language 'en')
-- Playing 'digits/4.gsm' (language 'en')
-- Executing [1234@outbound-ivr:2] Playback("DAHDI/1-1",
"demo-congrats") in new stack
-- Playing 'demo-congrats.gsm' (language 'en')
-- Executing [1234@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
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Re: [asterisk-users] Orginate not working well with PSTN lines
Dear all,
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 ?
Thanks ,
Ashik
On Tue, Apr 26, 2011 at 4:28 PM, Jim Dickenson wrote:
> "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:dicken...@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@outbound-ivr:1] SayDigits("DAHDI/1-1", "1234")
>> in new stack
>> -- Playing 'digits/1.gsm' (language 'en')
>> -- Playing 'digits/2.gsm' (language 'en')
>> -- Playing 'digits/3.gsm' (language 'en')
>> -- Playing 'digits/4.gsm' (language 'en')
>> -- Executing [1234@outbound-ivr:2] Playback("DAHDI/1-1",
>> "demo-congrats") in new stack
>> -- Playing 'demo-congrats.gsm' (language 'en')
>> -- Executing [1234@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
>
>
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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, Apr 27, 2011 at 12:44 PM, Gilles wrote: > On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali > 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 > -- _ -- 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] Orginate not working well with PSTN lines
I thank everyone, for their fruitfull informations. Regards, Ashik Ali On Fri, Apr 29, 2011 at 2:04 AM, Gilles wrote: > On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali > 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 > -- _ -- 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] agi playback to execute say.conf settings
Hi all, I am able to understand your solutions. Depending upon the india number reading method, I changed number reading setting in say.conf language. For more details visit my blog http://asterisknumbertovoice.blogspot.com/. It is working well with playback(num:123456,say) when I specified it in dialplan. Thanks, Ashik On Thu, Sep 2, 2010 at 7:08 PM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *asteriskguru > asteriskguru > *Subject:* [asterisk-users] agi playback to execute say.conf settings > > > > >Hi all, > > >I am using asterisk-1.6.2.10. I changed say.conf script for customized > number reading. > > > > >but when I write it in agi does not working. Here is agi debug output > from asterisk. > > >AGI Rx << EXEC playback "num:333456,say" > -- AGI Script Executing Application: (playback) Options: > (num:333456,say) > >AGI Tx >> 200 result=0 > > > >Anybody have any ideas to work it out in agi playback ? > > Replace playback “num:334456,say” with “say number 334456” > > Refer to > > http://www.voip-info.org/wiki/view/say+number > > > > -- > _ > -- 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] agi playback to execute say.conf settings
hi, any response ? thanks, Ashik On Mon, Sep 6, 2010 at 12:01 PM, Ashik Ali wrote: > Hi all, > > I am able to understand your solutions. Depending upon the india number > reading method, I changed number reading setting in say.conf language. For > more details visit my blog http://asterisknumbertovoice.blogspot.com/. > > > It is working well with playback(num:123456,say) when I specified it in > dialplan. > > Thanks, > Ashik > > > On Thu, Sep 2, 2010 at 7:08 PM, Danny Nicholas wrote: > >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asteriskguru >> asteriskguru >> *Subject:* [asterisk-users] agi playback to execute say.conf settings >> >> >> >> >Hi all, >> >> >I am using asterisk-1.6.2.10. I changed say.conf script for customized >> number reading. >> >> >> >> >but when I write it in agi does not working. Here is agi debug output >> from asterisk. >> >> >AGI Rx << EXEC playback "num:333456,say" >> -- AGI Script Executing Application: (playback) Options: >> (num:333456,say) >> >AGI Tx >> 200 result=0 >> >> >> >Anybody have any ideas to work it out in agi playback ? >> >> Replace playback “num:334456,say” with “say number 334456” >> >> Refer to >> >> http://www.voip-info.org/wiki/view/say+number >> >> >> >> -- >> _ >> -- 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] agi playback to execute say.conf settings
Hi danny,
Shall we take it as agi bug ?
Thanks,
Ashik
On Thu, Sep 9, 2010 at 7:10 PM, Danny Nicholas wrote:
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ashik Ali
> *Sent:* Thursday, September 09, 2010 2:06 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] agi playback to execute say.conf settings
>
>
>
> hi,
>
> any response ?
>
> thanks,
> Ashik
>
>
> exec playback in AGI expects to find a file or set of files in
> /var/lib/asterisk/sounds. Here is the vital snippet from the PERL AGI I
> whipped up to test your query
>
> print "EXEC PLAYBACK beep \"\"\n";
>
> #print "EXEC PLAYBACK 'num:334456,say' \"\"\n"; this line will fail with
>
> -- AGI Script Executing Application: (PLAYBACK) Options:
> ('num:334456,say')
>
> [Sep 9 08:33:16] WARNING[8569]: file.c:664 ast_openstream_full: File
> 'num:334456,say' does not exist in any format
>
> [Sep 9 08:33:16] WARNING[8569]: file.c:991 ast_streamfile: Unable to open
> 'num:334456,say' (format 0x4 (ulaw)): No such file or directory
>
> [Sep 9 08:33:16] WARNING[8569]: app_playback.c:440 playback_exec:
> ast_streamfile failed on SIP/134-000a for 'num:334456,say'
>
> print "SAY NUMBER 334456 \"\"\n"; This does the same thing in AGI as
> playback(num:334456,say)
>
> print "EXEC PLAYBACK beep \"\"\n";
>
> my $result = ;
>
> --
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Re: [asterisk-users] agi playback to execute say.conf settings
Hi danny, U r the one responding for me. Thanks a lot. How do we make it visible to asterisk developers. Thanks, Ashik On Tue, Sep 14, 2010 at 7:30 PM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ashik Ali > *Sent:* Tuesday, September 14, 2010 2:27 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] agi playback to execute say.conf settings > > > > Hi danny, > > > Shall we take it as agi bug ? > > Thanks, > Ashik > > > For lack of a better description, yes. In truth, it is more of an > “inconsistency” than an actual “bug” > > -- > _ > -- 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
