Re: [asterisk-users] Asterisk Test Suite error
- Original Message - > From: "upendra" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, September 18, 2012 12:30:04 AM > Subject: Re: [asterisk-users] Asterisk Test Suite error > Hi Matthew , > i have enabled the framework and tested the script, after running i > am getting some FAILS > -> tests/channels/SIP/refer_replaces_to_self --- FAILED > --> tests/channels/SIP/sip_tls_call --- FAILED > --> tests/channels/SIP/sip_cause --- FAILED > --> tests/masquerade --- FAILED > let me know still what i am missing in the testsuite. So, as I explained earlier, the Asterisk Test Suite will tell you what dependencies you are missing. If a dependency is missing, it will skip the test and not execute it. Since the tests appear to have executed and failed, it isn't a dependency problem. So you probably aren't "missing" anything in the Test Suite. When a test fails, it will provide you with the messages from the Test Suite log file (located in ./logs) of verbosity WARNING and higher. Those will often (but not always) contain the reasons for the test failure. Even then, more often than not, you have to inspect the Test Suite logs and sometimes the archived Asterisk logs to determine why the test failed. As much as I'd love to say I'll debug your errors for you, the fact is that some of those tests are rather complex and debugging failures in them can take some significant effort. For example, the masquerade test creates 300 Local channels and collapses them all down through optimization. Finding the problem when that test fails is a non-trivial effort. (And these tests do pass on the current Bamboo build agent, as well as on my development machine. So off the top of my head, I don't know why they would be failing on your machine.) The Asterisk Test Suite is a tool to aid in Asterisk development and test. If you don't feel comfortable debugging problems in Asterisk, then it might not be the tool for you. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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] chan_mobile
On Tue, 2012-09-18 at 17:43 +0100, Sebastian Arcus wrote: > Hi Hans, > > > The following page has some useful info: > > http://www.voip-info.org/wiki/view/chan_mobile > > Sebastian Indeed. Didn't realise it was so picky. just bought a couple of bt-adapters. Will try again tomorrow and feed the results into the wiki.. Tnx. -- _ -- 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] Trigger Asterisk after data inserted in mysql
On 9/18/2012 3:41 PM, Ahmed Munir wrote: Hi all, I would like to know, is there a way to trigger Asterisk after data inserted into mysql DB? Like here what I'm trying to do, when the new data inserted into MySQL DB, it sends the request to Asterisk along with the new data (that is inserted in DB) for making outbound call i.e. Realtime. Currently I've set a cron job that execute my script every 30 seconds and checks for a new data in DB. If new data is inserted in 30 seconds that script will run and sends the data to Asterisk for making calls. (This is the case which I'm thinking to avoid) Please advise. You could create a trigger in mysql that calls a shell script that pokes Asterisk properly. Look here for a start: http://forums.mysql.com/read.php?99,170973,236208#msg-236208 -- _ -- 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] Trigger Asterisk after data inserted in mysql
Hi all, I would like to know, is there a way to trigger Asterisk after data inserted into mysql DB? Like here what I'm trying to do, when the new data inserted into MySQL DB, it sends the request to Asterisk along with the new data (that is inserted in DB) for making outbound call i.e. Realtime. Currently I've set a cron job that execute my script every 30 seconds and checks for a new data in DB. If new data is inserted in 30 seconds that script will run and sends the data to Asterisk for making calls. (This is the case which I'm thinking to avoid) Please advise. -- Regards, Ahmed Munir Chohan -- _ -- 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] chan_mobile
Hi Hans, On 18/09/12 08:04, Hans Witvliet wrote: Hi all, In one of my other project i had a look at chan_mobile. I build 1.8.15.1 with the apropiate module. (in my distro asterisk is build without chan_mobile ;-) After i filled in the mac-addresses of the BT-adapter and the one from my phone, i see it is recognized, got connected, and immediate gets disconnected. What phone and bluetooth adapter are you using? Some of the most compatible phones for this sort of stuff seem to be Nokia phones. I've used three different models so far with success - although they always disconnect from bluetooth at the end of the call. They will reconnect again after a while. See note at the bottom of the page linked below. When it comes to bluetooth adapters, the ones with Cambridge Silicon Radio (CSR) seem to be the best for this job. The following page has some useful info: http://www.voip-info.org/wiki/view/chan_mobile Sebastian -- _ -- 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] Trunk SCCP
Hi all. I compiled the module chan_sccp, now its possible deploy trunk SCCP with Callmanager? Anyone? Regards-- _ -- 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] Any workaround for res_speech_lumenvox.so issue?
You could go back to a version that it works in and apply patches to it. > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Richard Kenner > Sent: Tuesday, September 18, 2012 9:48 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Any workaround for res_speech_lumenvox.so issue? > > The latest version of res_speech_lumenvox.so doesn't seem to work and > nobody seems to know when a version that works will be available. It > looks to me like this is some sort of timeout issue. Does anybody > have a workaround to allow this to be used? (I know about UniMRCP, > but find it quite "heavy".) > > -- > _ > -- 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] Any workaround for res_speech_lumenvox.so issue?
The latest version of res_speech_lumenvox.so doesn't seem to work and nobody seems to know when a version that works will be available. It looks to me like this is some sort of timeout issue. Does anybody have a workaround to allow this to be used? (I know about UniMRCP, but find it quite "heavy".) -- _ -- 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 HANGUP PROBLEM
On Tuesday 18 September 2012, Mehdi Rahimi wrote: > Hi Tony, > > Thank you for your attention , and appreciate your contribution . > You are right we can not do anything till the caller hangup BUT how > can we prevent to hearing DTMF when someone else is trying on another > extension ? > to clearance : > someone calls (from landlines os mobile , no difference) and our AGI > has executed and after some processes finish and hangup , but the > caller has not hungup yet and till then if i pickup my extension and > try to call , that caller who has not hungup the call yet can hear > DTMF and that's a problem and some conflict. Yes, that is the way a "calling party disconnects" telephone network works. Someone else rang you --> You answered --> You put the phone down --> They didn't put their phone down --> You will still be connected to them --> If you dial, they will hear your digits; if you speak, they will hear your voice. It is a telephone company issue, not an Asterisk issue. Asterisk gets around this the only way it can: by not marking an FXO line on which an incoming call has been answered as "free" until the calling party has hung up. -- AJS Answers come *after* questions. -- _ -- 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 HANGUP PROBLEM
Hi Tony,
Thank you for your attention , and appreciate your contribution .
You are right we can not do anything till the caller hangup BUT how
can we prevent to hearing DTMF when someone else is trying on another
extension ?
to clearance :
someone calls (from landlines os mobile , no difference) and our AGI
has executed and after some processes finish and hangup , but the
caller has not hungup yet and till then if i pickup my extension and
try to call , that caller who has not hungup the call yet can hear
DTMF and that's a problem and some conflict.
Regards,
Mehdi
On Tue, Sep 18, 2012 at 5:35 PM, Tony Mountifield wrote:
> In article
> ,
> SamyGo wrote:
>>
>> So basically the FXO cards configurations need to be tweaked i.e
>> hanguponpolarityinverse=yes etc.
>> Since this is a Hangup request initiated by the SIP client, Asterisk then
>> atleast it should close all the media streams and channel should get
>> deleted.
>> Keeping an eye on BYE : *CLI> "sip set debug on" Then make this call and
>> see if a SIP BYE method is triggered properly and appears on screen.
>> More likely you need to look into you dahdi configs.
>>
>> Thanks,
>> Sammy
>
> I think you are misunderstanding the OP's issue.
>
> Hangup on polarity reversal would only apply if Asterisk were making the
> call to a phone and wanted to me informed if the phone (called party)
> hung up.
>
> The OP's situation is different. The extension below is invoked by an
> INCOMING call to Asterisk, and he is then trying to hang up that call
> from the Asterisk (called) end.
>
> If the caller is a SIP phone, that is fine, as either end can hang up.
>
> Hi problem is that when the incoming call is via his FXO port, the PSTN
> does not drop the call when the Asterisk end hangs up the FXO line. In
> this scenario there is on SIP involved. The problem is that the PSTN
> will not drop the call when the called party on an analogue line hangs
> up, until after a long timeout. There is usually no solution to this.
>
> Cheers
> Tony
>
>> On Tue, Sep 18, 2012 at 2:03 PM, Tony Mountifield wrote:
>>
>> > In article <
>> > caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com>,
>> > Mehdi Rahimi wrote:
>> > > Hi all,
>> > >
>> > > I need to handle a problem from AGI please guide me
>> > >
>> > > in extensions_custom.conf :
>> > >
>> > > exten => s,1,Answer
>> > > exten => s,n,AGI(hang.php)
>> > > exten => s,n,Hangup
>> > >
>> > > in hang.php :
>> > >
>> > > #!/usr/bin/php -q
>> > > > > > set_time_limit(30);
>> > > require('phpagi.php');
>> > > error_reporting(E_ALL);
>> > > $agi = new AGI();
>> > > $agi->answer();
>> > > $agi->say_number('1');
>> > > $agi->hangup();
>> > > ?>
>> > >
>> > >
>> > > calling from an extension has no problem but whenever i use landline
>> > > or mobile it can not hangup the call and the caller has to hangup the
>> > > call.
>> >
>> > In the UK phone network, and I suspect in many other countries too, for
>> > analogue lines it is the caller who holds the call open. For example in
>> > a call between two normal analogue phones, the called party can hangup
>> > their phone, and then within a short while pick it up again (or another
>> > phone on the same line) and the caller is still there. Hanging up the
>> > called phone does not clear down the call until after quite a long
>> > timeout (a couple of minutes perhaps).
>> >
>> > In your above example with Asterisk connected to an analogue line with an
>> > FXO card, Asterisk is the called party, and is therefore unable to clear
>> > down the line forcibly. This is not an Asterisk or AGI problem but a PSTN
>> > one.
>> >
>> > Cheers
>> > Tony
>> > --
>> > Tony Mountifield
>> > Work: t...@softins.co.uk - http://www.softins.co.uk
>> > Play: t...@mountifield.org - http://tony.mountifield.org
>> >
>> > --
>> > _
>> > -- 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
>> >
>>
>> -=-=-=-=-=-
>> [Alternative: text/html]
>> -=-=-=-=-=-
>> -=-=-=-=-=-
>>
>> --
>> _
>> -- 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
>> -=-=-=-=-=-
>
>
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
> --
> _
> -- Bandwidth and Col
Re: [asterisk-users] AGI HANGUP PROBLEM
In article ,
SamyGo wrote:
>
> So basically the FXO cards configurations need to be tweaked i.e
> hanguponpolarityinverse=yes etc.
> Since this is a Hangup request initiated by the SIP client, Asterisk then
> atleast it should close all the media streams and channel should get
> deleted.
> Keeping an eye on BYE : *CLI> "sip set debug on" Then make this call and
> see if a SIP BYE method is triggered properly and appears on screen.
> More likely you need to look into you dahdi configs.
>
> Thanks,
> Sammy
I think you are misunderstanding the OP's issue.
Hangup on polarity reversal would only apply if Asterisk were making the
call to a phone and wanted to me informed if the phone (called party)
hung up.
The OP's situation is different. The extension below is invoked by an
INCOMING call to Asterisk, and he is then trying to hang up that call
from the Asterisk (called) end.
If the caller is a SIP phone, that is fine, as either end can hang up.
Hi problem is that when the incoming call is via his FXO port, the PSTN
does not drop the call when the Asterisk end hangs up the FXO line. In
this scenario there is on SIP involved. The problem is that the PSTN
will not drop the call when the called party on an analogue line hangs
up, until after a long timeout. There is usually no solution to this.
Cheers
Tony
> On Tue, Sep 18, 2012 at 2:03 PM, Tony Mountifield wrote:
>
> > In article <
> > caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com>,
> > Mehdi Rahimi wrote:
> > > Hi all,
> > >
> > > I need to handle a problem from AGI please guide me
> > >
> > > in extensions_custom.conf :
> > >
> > > exten => s,1,Answer
> > > exten => s,n,AGI(hang.php)
> > > exten => s,n,Hangup
> > >
> > > in hang.php :
> > >
> > > #!/usr/bin/php -q
> > > > > set_time_limit(30);
> > > require('phpagi.php');
> > > error_reporting(E_ALL);
> > > $agi = new AGI();
> > > $agi->answer();
> > > $agi->say_number('1');
> > > $agi->hangup();
> > > ?>
> > >
> > >
> > > calling from an extension has no problem but whenever i use landline
> > > or mobile it can not hangup the call and the caller has to hangup the
> > > call.
> >
> > In the UK phone network, and I suspect in many other countries too, for
> > analogue lines it is the caller who holds the call open. For example in
> > a call between two normal analogue phones, the called party can hangup
> > their phone, and then within a short while pick it up again (or another
> > phone on the same line) and the caller is still there. Hanging up the
> > called phone does not clear down the call until after quite a long
> > timeout (a couple of minutes perhaps).
> >
> > In your above example with Asterisk connected to an analogue line with an
> > FXO card, Asterisk is the called party, and is therefore unable to clear
> > down the line forcibly. This is not an Asterisk or AGI problem but a PSTN
> > one.
> >
> > Cheers
> > Tony
> > --
> > Tony Mountifield
> > Work: t...@softins.co.uk - http://www.softins.co.uk
> > Play: t...@mountifield.org - http://tony.mountifield.org
> >
> > --
> > _
> > -- 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
> >
>
> -=-=-=-=-=-
> [Alternative: text/html]
> -=-=-=-=-=-
> -=-=-=-=-=-
>
> --
> _
> -- 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
> -=-=-=-=-=-
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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Re: [asterisk-users] Hangup not detected
On Tuesday 18 September 2012, Mehdi Rahimi wrote: > Hi AJS, > > Thank you for your reply , I am using this in IRAN so please guide me > what to do and and explain me more. > Look forward to hearing from your side. > Regards, > Mehdi Unfortunately I am not familiar with the Iranian telephone system. You might have to search for relevant technical standards documentation. For a start, try setting your location to UK -- and if it behaves a bit better, that will be your problem. -- AJS Answers come *after* questions. -- _ -- 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] Hangup not detected
Hello In indications.com are the tones for several countries On Sep 18, 2012 4:34 AM, "Mehdi Rahimi" wrote: > Hi AJS, > > Thank you for your reply , I am using this in IRAN so please guide me > what to do and and explain me more. > Look forward to hearing from your side. > Regards, > Mehdi > > On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles > wrote: > > On Tuesday 18 September 2012, Satria Anamarta wrote: > >> Hi, > >> I just realize in these few days there are many calls that already > hangup > >> but not detected by Asterisk. > >> Those calls occupy PSTN lines and need to be manually terminated through > >> Flash Operation Panel or phycally disconnect the PSTN lines. > >> This never happen before but as long as I can remember, there are no > change > >> in configuration. > >> > >> Any ideas how to solve this? > > > > If you are using analogue phone lines in some country that uses a > British- > > style telephone system (line wires called "A" and "B", not "tip" and > "ring"; > > polarity reversal before ringing; double ring on incoming call), then by > > design only the calling party can terminate a call once established. If > > someone rings you and you hang up but they stay on the line, you will > still be > > connected to them if you later pick up the phone -- the call is only > > disconnected once the calling party hangs up. > > > > Asterisk is aware of this, and takes steps to mitigate it. The fix is > simply > > to make sure you specify the correct country in your DAHDI configuration. > > > > -- > > AJS > > > > Answers come *after* questions. > > > > -- > > _ > > -- 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
Re: [asterisk-users] AGI HANGUP PROBLEM
Hi,
So basically the FXO cards configurations need to be tweaked i.e
hanguponpolarityinverse=yes etc.
Since this is a Hangup request initiated by the SIP client, Asterisk then
atleast it should close all the media streams and channel should get
deleted.
Keeping an eye on BYE : *CLI> "sip set debug on" Then make this call and
see if a SIP BYE method is triggered properly and appears on screen.
More likely you need to look into you dahdi configs.
Thanks,
Sammy
On Tue, Sep 18, 2012 at 2:03 PM, Tony Mountifield wrote:
> In article <
> caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com>,
> Mehdi Rahimi wrote:
> > Hi all,
> >
> > I need to handle a problem from AGI please guide me
> >
> > in extensions_custom.conf :
> >
> > exten => s,1,Answer
> > exten => s,n,AGI(hang.php)
> > exten => s,n,Hangup
> >
> > in hang.php :
> >
> > #!/usr/bin/php -q
> > > set_time_limit(30);
> > require('phpagi.php');
> > error_reporting(E_ALL);
> > $agi = new AGI();
> > $agi->answer();
> > $agi->say_number('1');
> > $agi->hangup();
> > ?>
> >
> >
> > calling from an extension has no problem but whenever i use landline
> > or mobile it can not hangup the call and the caller has to hangup the
> > call.
>
> In the UK phone network, and I suspect in many other countries too, for
> analogue lines it is the caller who holds the call open. For example in
> a call between two normal analogue phones, the called party can hangup
> their phone, and then within a short while pick it up again (or another
> phone on the same line) and the caller is still there. Hanging up the
> called phone does not clear down the call until after quite a long
> timeout (a couple of minutes perhaps).
>
> In your above example with Asterisk connected to an analogue line with an
> FXO card, Asterisk is the called party, and is therefore unable to clear
> down the line forcibly. This is not an Asterisk or AGI problem but a PSTN
> one.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
> --
> _
> -- 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:
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asterisk-users mailing list
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Re: [asterisk-users] AGI HANGUP PROBLEM
In article ,
Mehdi Rahimi wrote:
> Hi all,
>
> I need to handle a problem from AGI please guide me
>
> in extensions_custom.conf :
>
> exten => s,1,Answer
> exten => s,n,AGI(hang.php)
> exten => s,n,Hangup
>
> in hang.php :
>
> #!/usr/bin/php -q
>set_time_limit(30);
> require('phpagi.php');
> error_reporting(E_ALL);
> $agi = new AGI();
> $agi->answer();
> $agi->say_number('1');
> $agi->hangup();
> ?>
>
>
> calling from an extension has no problem but whenever i use landline
> or mobile it can not hangup the call and the caller has to hangup the
> call.
In the UK phone network, and I suspect in many other countries too, for
analogue lines it is the caller who holds the call open. For example in
a call between two normal analogue phones, the called party can hangup
their phone, and then within a short while pick it up again (or another
phone on the same line) and the caller is still there. Hanging up the
called phone does not clear down the call until after quite a long
timeout (a couple of minutes perhaps).
In your above example with Asterisk connected to an analogue line with an
FXO card, Asterisk is the called party, and is therefore unable to clear
down the line forcibly. This is not an Asterisk or AGI problem but a PSTN one.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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Re: [asterisk-users] AGI HANGUP PROBLEM
ِDear Sammy,
Thank you for your following ,
1- Land line i mean telco company which is calling to my server , i
use FXO VOIP CARD (ATCOM 4 port) and test on a gateway too.
2-please explain me more about "Enable SIP traces and keep an eye on
the originating BYE request"
Regards,
Mehdi
On Tue, Sep 18, 2012 at 12:01 PM, SamyGo wrote:
> Hi,
>
> Just following this thread for few days, I've some basic troubleshooting
> questions for you.
> 1- What do you mean by calling from landline? How is your Landline /mobile
> reaching your asterisk box ? is there a Hardware card ! or a VoIP provider.
> 2- Enable SIP traces and keep an eye on the originating BYE request when the
> agi->hangup() is called. See if that BYE reaches to the caller !
>
> I suspect its a Hardware card that is not dropping the channel and
> maintaining the call with server.
>
> Regards,
> Sammy
>
>
> On Tue, Sep 18, 2012 at 12:49 PM, Hoggins! wrote:
>>
>> Hello,
>>
>> I experience the same problem, and I would really appreciate if someone
>> could give us a hint on that.
>>
>> Hoggins!
>>
>> Le 17/09/2012 19:22, Mehdi Rahimi a écrit :
>> > Hi all,
>> >
>> > I need to handle a problem from AGI please guide me
>> >
>> > in extensions_custom.conf :
>> >
>> > exten => s,1,Answer
>> > exten => s,n,AGI(hang.php)
>> > exten => s,n,Hangup
>> >
>> > in hang.php :
>> >
>> > #!/usr/bin/php -q
>> > > > set_time_limit(30);
>> > require('phpagi.php');
>> > error_reporting(E_ALL);
>> > $agi = new AGI();
>> > $agi->answer();
>> > $agi->say_number('1');
>> > $agi->hangup();
>> > ?>
>> >
>> >
>> > calling from an extension has no problem but whenever i use landline
>> > or mobile it can not hangup the call and the caller has to hangup the
>> > call.
>> > if the caller does not hangup the call it becomes kind of SPY (the
>> > caller can listen DTMF if someone call from an extension)
>> >
>> > I am using elastix 2.3.0 which has asterisk 1.8.10.0 .
>> >
>> > I really appreciate your sharing.
>> >
>> > Regards,
>> > Mehdi
>> >
>> > --
>> > _
>> > -- 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
>
>
>
> --
> _
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Re: [asterisk-users] Hangup not detected
Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles wrote: > On Tuesday 18 September 2012, Satria Anamarta wrote: >> Hi, >> I just realize in these few days there are many calls that already hangup >> but not detected by Asterisk. >> Those calls occupy PSTN lines and need to be manually terminated through >> Flash Operation Panel or phycally disconnect the PSTN lines. >> This never happen before but as long as I can remember, there are no change >> in configuration. >> >> Any ideas how to solve this? > > If you are using analogue phone lines in some country that uses a British- > style telephone system (line wires called "A" and "B", not "tip" and "ring"; > polarity reversal before ringing; double ring on incoming call), then by > design only the calling party can terminate a call once established. If > someone rings you and you hang up but they stay on the line, you will still be > connected to them if you later pick up the phone -- the call is only > disconnected once the calling party hangs up. > > Asterisk is aware of this, and takes steps to mitigate it. The fix is simply > to make sure you specify the correct country in your DAHDI configuration. > > -- > AJS > > Answers come *after* questions. > > -- > _ > -- 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 HANGUP PROBLEM
Hi,
Just following this thread for few days, I've some basic troubleshooting
questions for you.
1- What do you mean by calling from landline? How is your Landline /mobile
reaching your asterisk box ? is there a Hardware card ! or a VoIP provider.
2- Enable SIP traces and keep an eye on the originating BYE request when
the agi->hangup() is called. See if that BYE reaches to the caller !
I suspect its a Hardware card that is not dropping the channel and
maintaining the call with server.
Regards,
Sammy
On Tue, Sep 18, 2012 at 12:49 PM, Hoggins! wrote:
> Hello,
>
> I experience the same problem, and I would really appreciate if someone
> could give us a hint on that.
>
> Hoggins!
>
> Le 17/09/2012 19:22, Mehdi Rahimi a écrit :
> > Hi all,
> >
> > I need to handle a problem from AGI please guide me
> >
> > in extensions_custom.conf :
> >
> > exten => s,1,Answer
> > exten => s,n,AGI(hang.php)
> > exten => s,n,Hangup
> >
> > in hang.php :
> >
> > #!/usr/bin/php -q
> > > set_time_limit(30);
> > require('phpagi.php');
> > error_reporting(E_ALL);
> > $agi = new AGI();
> > $agi->answer();
> > $agi->say_number('1');
> > $agi->hangup();
> > ?>
> >
> >
> > calling from an extension has no problem but whenever i use landline
> > or mobile it can not hangup the call and the caller has to hangup the
> > call.
> > if the caller does not hangup the call it becomes kind of SPY (the
> > caller can listen DTMF if someone call from an extension)
> >
> > I am using elastix 2.3.0 which has asterisk 1.8.10.0 .
> >
> > I really appreciate your sharing.
> >
> > Regards,
> > Mehdi
> >
> > --
> > _
> > -- 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 --
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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] Hangup not detected
On Tuesday 18 September 2012, Satria Anamarta wrote: > Hi, > I just realize in these few days there are many calls that already hangup > but not detected by Asterisk. > Those calls occupy PSTN lines and need to be manually terminated through > Flash Operation Panel or phycally disconnect the PSTN lines. > This never happen before but as long as I can remember, there are no change > in configuration. > > Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called "A" and "B", not "tip" and "ring"; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- 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 HANGUP PROBLEM
Hello,
I experience the same problem, and I would really appreciate if someone
could give us a hint on that.
Hoggins!
Le 17/09/2012 19:22, Mehdi Rahimi a écrit :
> Hi all,
>
> I need to handle a problem from AGI please guide me
>
> in extensions_custom.conf :
>
> exten => s,1,Answer
> exten => s,n,AGI(hang.php)
> exten => s,n,Hangup
>
> in hang.php :
>
> #!/usr/bin/php -q
>set_time_limit(30);
> require('phpagi.php');
> error_reporting(E_ALL);
> $agi = new AGI();
> $agi->answer();
> $agi->say_number('1');
> $agi->hangup();
> ?>
>
>
> calling from an extension has no problem but whenever i use landline
> or mobile it can not hangup the call and the caller has to hangup the
> call.
> if the caller does not hangup the call it becomes kind of SPY (the
> caller can listen DTMF if someone call from an extension)
>
> I am using elastix 2.3.0 which has asterisk 1.8.10.0 .
>
> I really appreciate your sharing.
>
> Regards,
> Mehdi
>
> --
> _
> -- 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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[asterisk-users] chan_mobile
Hi all, In one of my other project i had a look at chan_mobile. I build 1.8.15.1 with the apropiate module. (in my distro asterisk is build without chan_mobile ;-) After i filled in the mac-addresses of the BT-adapter and the one from my phone, i see it is recognized, got connected, and immediate gets disconnected. Same behaviour if i use a completely different phone (BB). BT on either phone (one at the time) is constantly "on" and "visable'. Distance between dongle and phne just some centimeters Any suggestions? Hans -- _ -- 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
