Thank u so much. mod local stream really works to play sound files from a local
directory sequentially. Now can i jump to a specific file skipping the others?
What should i use in dialplan? Have u any idea. Nothing is mentioned in the doc
of mod lacal stream about that.
thanks Brian,
thank you so much for useful reply, It works very well now :)...
-
msp
Brian West-3 wrote:
>
> You would use something like this sofia/profile/
> [EMAIL PROTECTED];transport=tls
>
> /b
>
> On Dec 5, 2008, at 9:31 AM, shehzad p wrote:
>
>>
>>
>> I am wondering how to setup tw
make current or install current svn on a different box.
/b
On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> Ideas? Am I doing something stupid or is tone_detect not just right
> here?
___
Freeswitch-users mailing list
Freeswitch-users@lists.fre
Also got it on 9579 as well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael S Collins
Sent: Friday, December 05, 2008 8:42 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call
That's a
It's working... thanks a lot
On Fri, Dec 5, 2008 at 2:27 PM, MEHDi CHAABOUNi
<[EMAIL PROTECTED]>wrote:
> I changed the parameter expire-seconds to 30. Now, I'm starting to see the
> register request in the console.
> I'll wait a couple of hours and get back to you guys.
>
> Thanks
>
>
> On Fri, D
Jon,
You should also be able to do a
'order hosts,bind' in /etc/hosts, no
On Dec 5, 2008, at 11:43 AM, Jon Bruel wrote:
For the configuration of a gateway I need to use a specific proxy
domain name before the server (Covergence SBC with a BroadWorks
Application Server behind) accepts
That's a pretty old rev. Any chance you could make current?
-MC
Sent from my iPhone
On Dec 5, 2008, at 5:09 PM, "Frank @ Impact" <[EMAIL PROTECTED]>
wrote:
> I tried your suggested test. Here is the business end of the
> extension
> I tried.
>
>
>
>
>
>
I tried your suggested test. Here is the business end of the extension
I tried.
but I always got DTMF1=false in the info dump.
I am using FS 9210
I have tried sending a call from my sip phone connected to an asterisk
server to FS
you have an older revision.
put sip: instead of just
I recommend you update and either will work.
On Fri, Dec 5, 2008 at 1:51 PM, Jon Bruel <[EMAIL PROTECTED]> wrote:
> Thanks Anthony. Using the parameters:
>
>
>
>
> Returns error 900, and a 'ngrep port port-number' indicates that its
Thanks Anthony. Using the parameters:
Returns error 900, and a 'ngrep port port-number' indicates that its doesn't
try to register at all. I have now let the server look at a local DNS where I
have added a "wrong" A-record. That solves the issue, but your solution would
be cleaner. The vers
I changed the parameter expire-seconds to 30. Now, I'm starting to see the
register request in the console.
I'll wait a couple of hours and get back to you guys.
Thanks
On Fri, Dec 5, 2008 at 2:05 PM, Brian West <[EMAIL PROTECTED]> wrote:
> But you don't see the invite hitting FreeSWITCH? And y
Can you hit F8 and capture the debug output when making a call?
That'll help us see what's going on.
-MC
On Fri, Dec 5, 2008 at 10:59 AM, MEHDi CHAABOUNi
<[EMAIL PROTECTED]> wrote:
> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junc
Doh! Brian is way ahead of me, as usual...
On Fri, Dec 5, 2008 at 11:05 AM, Brian West <[EMAIL PROTECTED]> wrote:
> But you don't see the invite hitting FreeSWITCH? And you're behind
> NAT? Make it register every 30 seconds instead of the default 3600
>
> /b
>
> On Dec 5, 2008, at 10:59 AM, MEHD
But you don't see the invite hitting FreeSWITCH? And you're behind
NAT? Make it register every 30 seconds instead of the default 3600
/b
On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls
Actually, i did not mean that the line is dropped during a call...
FS is configured to accept calls from the Junction Networks SIP trunk to
make an audio conference.
When I start FS and I dial the number all is working fine. But, if I wait
for a couple of minutes and then make my call I get an erro
This variable is to specifically document the protocol specific last status
cause.
So you can know what the status was when you got a BYE or final response to
invite in the case of sip.
That's all it's for.
On Fri, Dec 5, 2008 at 12:43 PM, Apostolos Pantsiopoulos <[EMAIL
PROTECTED]>wrote:
>
forget call_timout it's your best bet
it's depricated.
On Fri, Dec 5, 2008 at 12:38 PM, Gonzalo Servat <[EMAIL PROTECTED]> wrote:
> On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins <[EMAIL PROTECTED]>wrote:
>
>> On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
>> <[EMAIL PROTECTED]> wrote:
>> >
I tested it and it works fine but it got me thinking...
Is just a copy of the cause to the other leg the correct way
to do it? Couldn't the two call legs hang up with different causes?
Especially when I could override the cause before it got send
to the e.g. calling side using e.g. the hangup com
On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins <[EMAIL PROTECTED]> wrote:
> On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
> <[EMAIL PROTECTED]> wrote:
> > call_timeout is only used if you are bridging 2 channels where one or
> both
> > of them is still unanswered.
> >
> > what you want to use
Thanks to all for their answers.
1. to Michael Collins
> >> I will need some time to digest all of this. I have an a104 but I
> >> don't have a solaris system for testing. I will report back as soon
> >> as I can
We with impatience will wait for results of your tests. If there will
be any q
> > no solution. I have a similar problem, when calling Freeswitch from my
> > cell phone (via a SIP provider), sometimes DTMF is not recognized
>> The important thing to note is that when using
>> a SIP softphone (X-Lite) I have never had this problem, DTMF is
> So i guess that using latest vers
What is the hangup cause?
/b
On Dec 5, 2008, at 10:23 AM, mehdix wrote:
> Any Ideas?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freesw
I've got a problem with configuring a SIP trunk from Junction Networks with
FS: it only works for a few minutes then the line is dropped.
I tried Unlimitel with no problem.
Any Ideas?
Thanks
--
View this message in context:
http://www.nabble.com/Provider%3A-Junction-Networks-tp20859688p208596
On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> call_timeout is only used if you are bridging 2 channels where one or both
> of them is still unanswered.
>
> what you want to use is originate_timeout and forget about call_timeout
>
> you also have
> leg_timeout and le
job-uuid can be used to match the BACKGROUND_JOB event which will
have the output of the originate command in the body.
since you are using bgapi it goes asyncronous and must deliver the reply to
you via the event interface.
On Fri, Dec 5, 2008 at 11:50 AM, Michael Collins <[EMAIL PROTECTED]> wr
Peter,
thanks, I will ruminate on this and get back with you as soon as I can.
-MC
On Fri, Dec 5, 2008 at 9:08 AM, Peter P GMX <[EMAIL PROTECTED]> wrote:
> I am a step further, When I set the cid-name then I can access the data
> dring
> channel_outgoing
> channel_originate
> channel_progress
> c
On Fri, Dec 5, 2008 at 8:32 AM, Frank @ Impact <[EMAIL PROTECTED]> wrote:
> Looks like tone detect might do it. But..
>
> If so, What frequency would we use for particular keys?
>
http://en.wikipedia.org/wiki/DTMF#Keypad
> Will tone_Detect sniff both legs or would we just do both r and w on the
I am a step further, When I set the cid-name then I can access the data
dring
channel_outgoing
channel_originate
channel_progress
channel_answer
However setting the caller_caller_id_number might be better.
This is the originate request:
freeswitch.api
bgapi
originate
{other_unique_id=ed525a3a-
set proxy to be the correct hostname and set register-proxy param to be the
correct IP
On Fri, Dec 5, 2008 at 10:43 AM, Jon Bruel <[EMAIL PROTECTED]> wrote:
> For the configuration of a gateway I need to use a specific proxy domain
> name before the server (Covergence SBC with a BroadWorks Appl
FreeSWITCH Version 1.0.trunk (10579)
Brian West wrote:
Did you say what SVN rev you're running.
/b
On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
Both legs are SIP. From non-registered endpoints (if of any use).
___
Freeswit
For the configuration of a gateway I need to use a specific proxy domain name
before the server (Covergence SBC with a BroadWorks Application Server behind)
accepts calls. The twist is that the right proxy name points the wrong
IP-address (the voicemail server for the account). I have tried to o
call_timeout is only used if you are bridging 2 channels where one or both
of them is still unanswered.
what you want to use is originate_timeout and forget about call_timeout
you also have
leg_timeout and leg_progress_timeout both to be set in the {}
that do the timeout from the perspective of t
It's easy enough to set the value on both legs try r10614
It was only set on the opposing leg before but since it's harmless to set it
on both i did it for you.
On Fri, Dec 5, 2008 at 10:23 AM, Brian West <[EMAIL PROTECTED]> wrote:
> Did you say what SVN rev you're running.
>
> /b
>
> On Dec 5,
Makes sense. I will look into this.
-MC
On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]>
wrote:
I am sending this second email to request/suggest/enquire about
something relevant :
Wouldn't it be useful to know which end of a specific call leg send
the protocol
s
Hello,
I have the same problem,
I don't understand the difference between
progress_timeout
originate_timeout
call_timeout.
I log timelimit_sec in switch_ivr_originate function and it seems,
if I set call_timeout then, timelimit_sec will be this value
if I set originate_timeout then timelimit_se
Looks like tone detect might do it. But..
If so, What frequency would we use for particular keys?
Will tone_Detect sniff both legs or would we just do both r and w on the
called leg?
Can the tone_Detect timeout just be a very large number or can we leave
out the timeout value so there is no t
The call should continue after FS hears the key press and responds with
its own key press tone. Then the call just continues on. They key
press from one of the parties would come some time after the call is
bridged. Maybe some 10 or 20 seconds into the call for example.
-Original Message-
Did you say what SVN rev you're running.
/b
On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
> Both legs are SIP. From non-registered endpoints (if of any use).
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http:/
Last I spoke to doug at sangoma, solaris support is still not in their
platform abstraction lib (there are drivers). Please contact sangoma
sales and request this.
Mike.
p.s. make sure to tell them its for FreeSWITCH
On Dec 5, 2008, at 11:12 AM, Brian West wrote:
> Does it list wanpipe TD
Will the call be terminated at that point or does it need to continue?
I do know that the tone_detect app can listen for a dtmf from either
direction and can trigger execution of another app/extension/etc.
However, I've never tried it on a bridged call, so I'm curious to see
what would happ
I am sending this second email to request/suggest/enquire about
something relevant :
Wouldn't it be useful to know which end of a specific call leg send the
protocol
specific hangup cause? Otherwise it would be difficult to understand
what really happened.
Michael S Collins wrote:
I will d
Both legs are SIP. From non-registered endpoints (if of any use).
Michael S Collins wrote:
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5, 2008, at 4:23 AM, Aposto
Does it list wanpipe TDM support on the Solaris builds of wanpipe? I
wasn't aware the TDM stuff was ported yet.
/b
On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:
> Evgeniy,
>
> I will need some time to digest all of this. I have an a104 but I
> don't have a solaris system for testing.
[EMAIL PROTECTED] wrote:
> After the tone is sent back out, we are done. There is
> nothing left to
> do.
Maybe you can look at:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
Ciao,
Claudio
Internet Email Confidentiality Footer
--
Yes. listen in for 1 DTMF during a call and then signal back a
different DTMF.
-Original Message-
From: [EMAIL PROTECTED] [mailto:freeswitch-
So receive DTMF respond with more DTMF?
/b
On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:
> After the tone is sent back out, we are done.
Evgeniy,
I will need some time to digest all of this. I have an a104 but I
don't have a solaris system for testing. I will report back as soon as
I can.
-MC
On Dec 5, 2008, at 6:53 AM, Evgeniy Zolotov <[EMAIL PROTECTED]> wrote:
> Greetings!
>
> Question about possibility of the use FreeS
So receive DTMF respond with more DTMF?
/b
On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:
> After the tone is sent back out, we are done. There is nothing left
> to
> do.
> No, this key press detection is during a bridged call between two
> parties. No IVR here. So, FS hears a key pres
What is your originate string?
-MC
On Dec 5, 2008, at 3:54 AM, Peter P GMX <[EMAIL PROTECTED]> wrote:
> I am building an IVR application where an incoming call is
> initiating an
> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> incoming channel) at originate time, I am a
After the tone is sent back out, we are done. There is nothing left to
do.
No, this key press detection is during a bridged call between two
parties. No IVR here. So, FS hears a key press tone during a call and
then responds to the parties with another/different key press tone.
-Origina
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]>
wrote:
> The proto_specific_hangup_cause is missing o
Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?
-Frank
___
Freeswitch-users
What would need to happen after the tone is sent back out? Also, would
this be part of something like an IVR?
-MC
On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <[EMAIL PROTECTED]>
wrote:
>
> Is there any dialplan instructions that could be added that would sit
> and listen during a call for
You would use something like this sofia/profile/
[EMAIL PROTECTED];transport=tls
/b
On Dec 5, 2008, at 9:31 AM, shehzad p wrote:
>
>
> I am wondering how to setup two freeswitch servers to route call
> with TLS
> configured between them.
___
Frees
Hello,
I have got some problems for the configuration of a simple conference which
should be established by calling an extension and automatically inviting 2
people.
Actually, this is based on the default configuration of Freeswitch
(extension 0911). I have changed it a little:
I am wondering how to setup two freeswitch servers to route call with TLS
configured between them.
As shown in wiki http://wiki.freeswitch.org/wiki/SIP_TLS, I created two
certificates on one freeswitch, and changed SIP profile by enabling tls in
it,
then Starting freeswitch it just opens port 50
Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?
___
Freeswitch-users maili
Check out mod_localstream on the wiki and see if that sounds like what
you need. I'm still learning it all myself but I believe that's where
you should start. Please report back with any questions and we will
take it from there!
-MC
On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <[EMAIL PROT
Greetings!
Question about possibility of the use FreeSWITCH for work with T1/E1
streams under Sun Solaris 10 a bit clears up (Solaris 11 is in condition
of alpha-version and not suitable for the industrial use). But answers
carry more negative sense.
Start of T1/E1 under Sun Solaris has 2 sta
On Dec 5, 2008, at 5:54 AM, Dennis wrote:
> 2008/12/5 Steve Underwood <[EMAIL PROTECTED]>:
>>> 1.) there is one error, we get always - no matter, if the fax was
>>> sent
>>> successfully or not.
>>> in the pastebin under http://pastebin.freeswitch.org/6338 you can
>>> see
>>> the error in the
On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:
> Hi Micheal,
>
> Thanks for the reply! cant I try with tone detect?
>
> Like dial a number in session and try to detect with tone detect
> and then bridge the call with some extension.
If you know the exact frequency of the tone you ca
On Dec 5, 2008, at 6:08 AM, Jan Kubr wrote:
> Hi,
> recently someone was mentioning an issue with DTMF here, but there was
> no solution. I have a similar problem, when calling Freeswitch from my
> cell phone (via a SIP provider), sometimes DTMF is not recognized
> (read app doesn't terminate). I
On Dec 5, 2008, at 6:54 AM, Peter P GMX wrote:
> I am building an IVR application where an incoming call is
> initiating an
> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> incoming channel) at originate time, I am able to reference to the
> incomig call, once the outgoin
I had some issues with some previous versions of FS , in trunk looks
that is fixed. ( Notice current svn revision is 10609 )
in sip profiles i have :
...
...
As codecs g711 ULAW (PCMU):
in vars.xml.conf :
So i guess that using latest version with a few changes in your config
should wo
The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this nornal?
And another question : what piece of info could inform me about who
hanged up?
--
--
I am building an IVR application where an incoming call is initiating an
outgoing call. When I pass a "variable_other_uuid" (the uuid of the
incoming channel) at originate time, I am able to reference to the
incomig call, once the outgoing call is set up. So the outgoing call can
see the uuid of th
Its not without music on hold completely. Say, e.g, moh is being played
but when i press 1 it should start playing files contained in a
specific directory sequentially or randomly. I havent got any solution to this
problem yet. Can anyone plz guide me to some documentation or anything else
regard
Hi Micheal,
Thanks for the reply! cant I try with tone detect?
Like dial a number in session and try to detect with tone detect and then
bridge the call with some extension.
--
Thank you with regards,
Gopal,
___
Freeswitch-users mailing list
Frees
Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is
2008/12/5 Steve Underwood <[EMAIL PROTECTED]>:
>> 1.) there is one error, we get always - no matter, if the fax was sent
>> successfully or not.
>> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
>> the error in the last line.
>> this is the full log of a fax in fs console log
Hi all,
I have come across a strange problem when using the phrases in conf/lang/en.
Initially I had a problem where FreeSwitch wouldn't load new subdirectories,
even when I included their paths in the en.xml file. I went ahead writing
all the phrases (I only have 3 so far) in en.xml but now it do
Thanks, you're right it seems to be an odbc problem, 64bit / 32bit clash I
think.
Joe
2008/12/2 Michael Jerris <[EMAIL PROTECTED]>
> Yes, it should work fine. As the error message says it didn't find
> the data source name you specified. You need to setup your odbc data
> source on the system
Its not without music on hold completely. Say, e.g, moh is being played but
when i press 1 it should start playing files contained in a specific directory
sequentially or randomly. Hope i m able to explain.
Faisal
--- On Fri, 12/5/08, Michael Col
Is this for Music on Hold? Or is it for a different application altogether?
Thanks,
MC
On Fri, Dec 5, 2008 at 12:37 AM, Faisal Maqsoodi
<[EMAIL PROTECTED]> wrote:
> Hi,
> Can i accomplish folder tasks with freeswitch? For instance, i need to play
> all sound files contained in a directory sequent
Hi,
Can i accomplish folder tasks with freeswitch? For instance, i need to play
all sound files contained in a directory sequentially or randomly. Plz help me
doing this.
Faisal
__
Check it out:
http://digg.com/software/FreeSWITCH_knocks_Asterisk_s_block_off
Please diggit left and right!!
-MC
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUB
I still know some folk in the 900-number business. They won't do
anything without
profanity in it ;-)
--Dave
Actually GM Voices won't do anything with profanity in it.
/b
On Dec 4, 2008, at 11:55 AM, Michael Collins wrote:
I think GM Voices levies a "naughtiness surcharge" but I'll see
76 matches
Mail list logo
