Something like the following perhaps??? Is this
possible?
This would be a bridge without answer would it not?
http://www.worldlink.net/sip_signals_b2bua.gif
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
On Behalf Of Kareem Hamdy
Sent: Wednesday, January 21, 2009 11:15 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to bridge without Answer? (Anthony
Minessale)
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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Sent: Wednesday, January 21, 2009 6:54 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 31, Issue 125
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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@gmail.com>
Subject: Re: [Freeswitch-users] How to bridge without
Answer?
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<191c3a030901210556n2d443179n17d8bbb9ed24b8ab@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@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.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users@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@hotmail.com
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@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@freeswitch.org>
Subject: Re: [Freeswitch-users] firing events from
javascript -
working
example needed
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<87f2f3b90901210603i39db5167rbc255cc78880a121@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@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
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
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------------------------------
Message: 3
Date: Wed, 21 Jan 2009 15:33:13 +0100
From: Krzysztof Zimnicki <krzysiez@go2.pl>
Subject: Re: [Freeswitch-users] Problem with digium
te220p
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<4c5d42470901210633h5f9abca0u6eb097c52c82987d@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@freeswitch.org> wrote:
> can you post your openzap.conf file?
> -MC
>
> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki
<krzysiez@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@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@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@freeswitch.org>
Subject: Re: [Freeswitch-users] Problem with digium
te220p
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<87f2f3b90901210645q773c8a82p4897843fb3c05699@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@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@freeswitch.org> wrote:
>>
>> can you post your openzap.conf file?
>> -MC
>>
>> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof
Zimnicki <krzysiez@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@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@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@lists.freeswitch.org
>
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>
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> http://www.freeswitch.org
>
>
------------------------------
Message: 5
Date: Wed, 21 Jan 2009 06:52:57 -0800
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] Hang up not received
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<87f2f3b90901210652m14abed25l5749e9792d0501f1@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis
<scott.ellis@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@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@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@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@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@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@hotmail.com
>>>>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888@conference.freeswitch.org
>>>> iax:guest@conference.freeswitch.org/888
>>>>
googletalk:conf+888@conference.freeswitch.org
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users@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@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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>
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>
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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@gmail.com>
Subject: [Freeswitch-users] ATA-answering machine
question/recommendation
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@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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