VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
Kareem.Hamdy at trustv...
Guest
|
 Posted: Wed Jan 21, 2009 11:29 am Post subject: [Freeswitch-users] How to bridge without Answer? (Anthony Mi |
 |
|
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-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request@lists.freeswitch.org
Sent: Wednesday, January 21, 2009 6:54 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 31, Issue 125
Send Freeswitch-users mailing list submissions to
freeswitch-users@lists.freeswitch.org
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request@lists.freeswitch.org
You can reach the person managing the list at
freeswitch-users-owner@lists.freeswitch.org
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Freeswitch-users digest..."
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:
| Quote: |
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/32390e36/attachment-0001.html
------------------------------
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:
------------------------------
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:
| Quote: | can you post your openzap.conf file?
-MC
On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki <krzysiez@go2.pl>
wrote:
| Quote: | | Quote: | Can you join irc later today? I will be on as mercutioviz. I would
like to discuss this more.
-MC
|
| Quote: | Sent from my iPhone
|
Sorry, i can't join to irc. Can you put your questions here? I'll try to
|
answer.
| Quote: |
Our CallCenter have strange situation, because now is working on Asterisk
|
and we can only put this card in other machine after 22 pm.
_______________________________________________
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
|
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/2a159f00/attachment-0001.html
------------------------------
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:
| Quote: | conf/openzap.conf
[span zt]
|
[span zt PRI_1]
| Quote: | name => OpenZAP
number => 1
trunk_type => E1
b-channel => 1-15
d-channel => 16
b-channel => 17-31
[span zt]
|
[span zt PRI_2]
------------------------------
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:
| Quote: | I had a similar problem, you can use
<action application="set" data="ringback=${au-ring}"/> (I added an "au"
ring definition to my vars.xml file)
To get what you want.
I also had a problem that you get two rings, then an answer then to the
system generated ring tone, which was confusing some of our (not to bright)
callers.
As we don't use callerID I turned that flag off in the openzap.conf.xml file
- I thought that this would do what I wanted (answer the instant the call is
detected), but the change in the config file does not make it all the way
down to the point where it takes action. At this point I hacked the code to
get what I wanted. I have to create a JIRA entry with the details yet.
As far as I understand, this is the right place for OpenZap, as it is a
product of the FS project.
|
At this point there is not a separate mailing list for OpenZAP stuff
so here is as good a place as any to ask OZ questions. 
-MC
| Quote: |
Scott
Tom?s wrote:
Scott, I imagined that it could be an OpenZap problem, but I didn't find an
OpenZap mailing list, so I sent the email to FS list. Do you know where can
I find more information about OpenZap hardware support and developement
status (I have special interest in Loop Start)??
Anthony and Ognjen, I've tried tone detection and thanks to that FS is
detecting hung up, but I faced the problem that tone detector answer the
call...
That's my dialplan:
<extension name="extension_name">
<condition field="destination_number" expression="^919999999$">
<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:
| Quote: |
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:
| Quote: |
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:
| Quote: |
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:
| Quote: |
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
|
________________________________
_______________________________________________
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
|
------------------------------
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/36129ec2/attachment.html
------------------------------
_______________________________________________
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
End of Freeswitch-users Digest, Vol 31, Issue 125
*************************************************
_______________________________________________
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 |
|
| Back to top |
|
 |
brian at freeswitch.org
Guest
|
| Posted: Wed Jan 21, 2009 11:39 am Post subject: [Freeswitch-users] How to bridge without Answer? (Anthony Mi |
|
|
You can already do this ... its how a phone call already works.
CALL A -> FS -> CALL B
Call A will answer when Call B is picked up passing the answer over to
Call A.
/b
On Jan 21, 2009, at 10:14 AM, Kareem Hamdy wrote:
| Quote: | 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
|
_______________________________________________
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 |
|
| Back to top |
|
|
mike at jerris.com
Guest
|
| Posted: Wed Jan 21, 2009 12:11 pm Post subject: [Freeswitch-users] How to bridge without Answer? (Anthony Mi |
|
|
a normal call should flow like that, with the possible exception of the ack handling, we don't wait for the a leg ack before we ack the b leg and the same for the 200ok to bye going the other way.
Mike
On Jan 21, 2009, at 11:44 AM, Adam Long wrote:
|
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
