[Freeswitch-users] How to bridge without Answer? (Anthony Minessale)
Kareem Hamdy
Kareem.Hamdy at trustvesta.com
Wed Jan 21 08:14:54 PST 2009
Hello everyone:
I think what Anthony wants is (please excuse me if I'm wrong - but what I'm assuming is) a call to come in - let's say that its DID goes to person A. He wants to ring person A, let person A pick up, and then bridge the call.
When working at an Asterisk VoIP vendor, I had a call in which a gentleman wanted just that. I think they paid for incoming calls or something.
Anthony, please let us know if that's accurate.
Thanks,
Kareem
-----Original Message-----
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Today's Topics:
1. Re: How to bridge without Answer? (Anthony Minessale)
2. Re: firing events from javascript - working example needed
(Michael Collins)
3. Re: Problem with digium te220p (Krzysztof Zimnicki)
4. Re: Problem with digium te220p (Michael Collins)
5. Re: Hang up not received (Michael Collins)
6. ATA-answering machine question/recommendation
(jonathan augenstine)
----------------------------------------------------------------------
Message: 1
Date: Wed, 21 Jan 2009 07:56:51 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Subject: Re: [Freeswitch-users] How to bridge without Answer?
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<191c3a030901210556n2d443179n17d8bbb9ed24b8ab at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
You can't.
Why would you need that? Are you trying to forward inbound calls from the
pstn to an ivr without answering them?
That could get you in trouble FYI.
On Wed, Jan 21, 2009 at 7:40 AM, shehzad p <pmhshz at gmail.com> wrote:
>
> Hi all,
>
> When I dial a number from Originator Gateway, It will route to Freeswitch
> Server and then FS will bridge the call to Terminator Gateway as below.
> Terminator Answer the call (and runs playback, and look for DTMF).
>
> |Originator Gateway|---------------> |FreeSwitch |------------------>
> |Terminator Gateway|
>
> I used bridge application to route call to Terminator.
> But my requirement is that when Terminator answer the call (Respnd with
> 200OK) , Freeswitch should NOT Answer call for A leg (Originater Gateway).
>
> How can be this done?
>
> Thanks in advance.
> msp.
> --
> View this message in context:
> http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21583334.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
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> Freeswitch-users at lists.freeswitch.org
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>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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Message: 2
Date: Wed, 21 Jan 2009 06:03:21 -0800
From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] firing events from javascript -
working example needed
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<87f2f3b90901210603i39db5167rbc255cc78880a121 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
can you create a pastebin with the two scripts in question? We'll take
a look and see if we can figure out what's going on.
Thanks,
MC
On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby <stevecrozz at gmail.com> wrote:
> I noticed the wiki has an example of sending a custom event from
> javascript: http://wiki.freeswitch.org/wiki/Javascript_Event , but I
> can't make it work. It doesn't fail or cause an error. But I never see
> an event on my listener script. Can someone confirm that this example
> does in fact work? or provide me with one that does?
>
> --Stephen
>
> _______________________________________________
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> Freeswitch-users at lists.freeswitch.org
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------------------------------
Message: 3
Date: Wed, 21 Jan 2009 15:33:13 +0100
From: Krzysztof Zimnicki <krzysiez at go2.pl>
Subject: Re: [Freeswitch-users] Problem with digium te220p
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<4c5d42470901210633h5f9abca0u6eb097c52c82987d at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
conf/openzap.conf
[span zt]
name => OpenZAP
number => 1
trunk_type => E1
b-channel => 1-15
d-channel => 16
b-channel => 17-31
[span zt]
name => OpenZAP
number => 2
trunk_type => E1
b-channel => 32-46
d-channel => 47
b-channel => 48-62
On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins <msc at freeswitch.org> wrote:
> can you post your openzap.conf file?
> -MC
>
> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki <krzysiez at go2.pl>
> wrote:
> >>Can you join irc later today? I will be on as mercutioviz. I would
> >>like to discuss this more.
> >>
> >>-MC
> >
> >>Sent from my iPhone
> >
> > Sorry, i can't join to irc. Can you put your questions here? I'll try to
> answer.
> >
> > Our CallCenter have strange situation, because now is working on Asterisk
> and we can only put this card in other machine after 22 pm.
> >
> > Thanks.
> >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Message: 4
Date: Wed, 21 Jan 2009 06:45:06 -0800
From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Problem with digium te220p
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<87f2f3b90901210645q773c8a82p4897843fb3c05699 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Okay, try the changes I note below
-MC
On Wed, Jan 21, 2009 at 6:33 AM, Krzysztof Zimnicki <krzysiez at go2.pl> wrote:
> conf/openzap.conf
>
> [span zt]
[span zt PRI_1]
> name => OpenZAP
> number => 1
> trunk_type => E1
> b-channel => 1-15
> d-channel => 16
> b-channel => 17-31
> [span zt]
[span zt PRI_2]
> name => OpenZAP
> number => 2
>
> trunk_type => E1
> b-channel => 32-46
> d-channel => 47
> b-channel => 48-62
>
>
> On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> can you post your openzap.conf file?
>> -MC
>>
>> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki <krzysiez at go2.pl>
>> wrote:
>> >>Can you join irc later today? I will be on as mercutioviz. I would
>> >>like to discuss this more.
>> >>
>> >>-MC
>> >
>> >>Sent from my iPhone
>> >
>> > Sorry, i can't join to irc. Can you put your questions here? I'll try to
>> > answer.
>> >
>> > Our CallCenter have strange situation, because now is working on
>> > Asterisk and we can only put this card in other machine after 22 pm.
>> >
>> > Thanks.
>> >
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
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> http://www.freeswitch.org
>
>
------------------------------
Message: 5
Date: Wed, 21 Jan 2009 06:52:57 -0800
From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Hang up not received
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<87f2f3b90901210652m14abed25l5749e9792d0501f1 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis <scott.ellis at novatex.com.au> wrote:
> I had a similar problem, you can use
> <action application="set" data="ringback=${au-ring}"/> (I added an "au"
> ring definition to my vars.xml file)
>
> To get what you want.
>
> I also had a problem that you get two rings, then an answer then to the
> system generated ring tone, which was confusing some of our (not to bright)
> callers.
>
> As we don't use callerID I turned that flag off in the openzap.conf.xml file
> - I thought that this would do what I wanted (answer the instant the call is
> detected), but the change in the config file does not make it all the way
> down to the point where it takes action. At this point I hacked the code to
> get what I wanted. I have to create a JIRA entry with the details yet.
>
> As far as I understand, this is the right place for OpenZap, as it is a
> product of the FS project.
At this point there is not a separate mailing list for OpenZAP stuff
so here is as good a place as any to ask OZ questions. :)
-MC
>
> Scott
>
> Tom?s wrote:
>
> Scott, I imagined that it could be an OpenZap problem, but I didn't find an
> OpenZap mailing list, so I sent the email to FS list. Do you know where can
> I find more information about OpenZap hardware support and developement
> status (I have special interest in Loop Start)??
>
> Anthony and Ognjen, I've tried tone detection and thanks to that FS is
> detecting hung up, but I faced the problem that tone detector answer the
> call...
>
> That's my dialplan:
>
> <extension name="extension_name">
> <condition field="destination_number" expression="^919999999$">
> <action application="tone_detect" data="busy 425,0 r +100 hangup 16
> 4"/>
> <action application="bridge"
> data="sofia/internal/1003%${server-domain-name},
> sofia/internal/1004%${server-domain-name}"/>
> </condition>
> </extension>
>
> When I receive a call from PSTN, tone detection answer the call (the caller
> hears only one first tone and then hears "nothing" until I pick up the call
> on softphone).
>
> So, I think that tone detection solution does not resolve my problem... Is
> there any other possibility to detect hang up without answering the call
> (using Loop Start signaling) or have we to wait until OpenZap is completely
> developed?
>
> Thanks in advance.
>
> On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <oseslija at gmail.com> wrote:
>>
>> Ok, as discussed with Tony on IRC channel I followed his directions which
>> lead to a successfull outcome (like it always does I might add :).
>>
>> One has to use tone_detect app in FreeSWITCH dialplan in order to check
>> for busy tones coming from the PSTN side and if matched fire a hangup
>> application. This is the snippet of my test dp that does the trick (from
>> extension Local_extensions in default.xml):
>>
>> <anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16
>> 4"/>
>> <anti-action application="bridge"
>> data="user/${dialed_extension}@${domain_name}"/>
>> This means that FS will listen to freq of 425 Hz and wait for 4 positive
>> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425
>> Hz is the freq telco here uses; for other countries I suggest getting the
>> ITU world tones pdf file and check there):
>>
>> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
>> TONE busy HIT 1/4
>> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
>> TONE busy HIT 2/4
>> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
>> TONE busy HIT 3/4
>> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
>> TONE busy HIT 4/4
>> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback()
>> TONE busy DETECTED
>> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup
>> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>> Regards,
>> Ognjen
>>
>> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija at gmail.com>
>> wrote:
>>>
>>> I tried similar setup with my analog card (X100P) and I'm having same
>>> issue. Call is not hungup on the oz side once the caller ends. My telco
>>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck
>>> to detecting busy tone from the telco side. I'll try to modify tones.conf
>>> accordingly.
>>>
>>> Regards,
>>> Ognjen
>>> (sekil)
>>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> This is a common issue with analog phones even traditional answering
>>>> machines suffer from it.
>>>> I'm sure you must have had an answering machine at some point that has
>>>> dial tone as the message it receives.
>>>>
>>>> Unless FreeSWITCH has some hint that the call has hungup it will not
>>>> stop trying to complete the call.
>>>>
>>>> If the other side is sending a busy tone to indicate hangup it's
>>>> possible to use the tone_detect app to pick
>>>> up on the tones and abort the call.
>>>>
>>>> Another thing you could do if you have unlimited inbound is explicitly
>>>> answer the call in the dialplan before
>>>> you call your sip phones this will give you a more profound hangup
>>>> detection but it will make every call count
>>>> even when nobody answers.
>>>>
>>>>
>>>>
>>>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s <tomasborrella at gmail.com> wrote:
>>>>>
>>>>> Hi all,
>>>>>
>>>>> I'm configuring my home PBX using FreeSwitch. I'm using a X101P card
>>>>> configured as FXO (conected to analog PSTN line) and I have several IP
>>>>> phones and softphones conected to FreeSwitch.
>>>>>
>>>>> I can call from an IP phone to other IP phone (the same with the
>>>>> softphones) and also from an IP phone (or softphone) to an external number
>>>>> thought PSTN.
>>>>>
>>>>> When I call from an external analog phone to FreeSwitch, I bridge the
>>>>> call to all internal IP phones and softphones and they ring, but the problem
>>>>> is that when I hang up the call in the external phone, all internal phones
>>>>> (IP phones and softphones) keeps ringing...
>>>>>
>>>>> I'm pretty sure the problem is that FreeSwitch don't receive the hang
>>>>> up, because I cann't see anything on the log.
>>>>>
>>>>> I've also created my own tones.conf for my country (Spain) but I'm not
>>>>> sure if it's ok (but I have the same problem with hang up)
>>>>>
>>>>> I've googled the list, and I've found several people with a similar
>>>>> problem but no solution...
>>>>>
>>>>> That's my pastebin with the most importants printouts and config files:
>>>>> http://pastebin.freeswitch.org/6822
>>>>>
>>>>> Thank you very much in advance.
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> _______________________________________________
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>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
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> http://www.freeswitch.org
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>
> _______________________________________________
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>
------------------------------
Message: 6
Date: Wed, 21 Jan 2009 06:53:41 -0800
From: jonathan augenstine <jaugenstine at gmail.com>
Subject: [Freeswitch-users] ATA-answering machine
question/recommendation
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I have an application that requires answering machine detection. I have not
been able to locate any documentation indicating that there is explicit
support for answering machine detection. I have received recommendations on
call flows that would include DTMF entry by the called party to detect by
implication answering machines, however, I need an explicit methodology. My
question is, does anyone have any experience with ATAs that might have this
capability. I am interested in any solution that might even include Avaya,
Cisco, or other hardware device interfaced with Freeswitch that would
provide an explicit answering machine detection capability.
Jonathan
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