some basic knowledge about sockets and
freeswitch. Could someone help to find the right way, please?
Thanks and kind regards
Dennis
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in a reversed order. playback five.wav, playback
fifty.wav, blayback twelve.wav. Because the first two files will be
paused, twelve.wav will be played first.
Is this the way I have to go or are there chances, that there is a
slicker solution for this?
Thanks for the help.
Dennis
. There is no
chance to stop playing the list, if I whish to. If a long list of
files is playing and an event accours, which makes me want to stop
playing all files, fs does not react on new commands, till all files
where played back - even break won't do anything.
Is there a solution for this?
Dennis
Ok, if there is nothing implemented yet, which can break playing
files, which where started with event-lock:true, I will go your
suggested way.
I think, that this would be a very helpful feature.
Kind regards
Dennis
2008/10/21 Anthony Minessale [EMAIL PROTECTED]:
yes post a feature request
sessions individually?
Sorry for all these questions, but we have serious problems, because
we can not find the answers.
Thanks
Dennis
2008/10/21 Anthony Minessale [EMAIL PROTECTED]:
once you are controlling both sessions individually use api to send the
uuid_bridge command:
api uuid_bridge uuid
2008/10/21 Anthony Minessale [EMAIL PROTECTED]:
api break uuid will break one of them just not all. it has nothing to do
with event lock
event lock means do not parse events recursively like the behavior you
described.
Ah, ok, now I understand what event lock means, thanks.
Since you asked
2008/10/21 Michael Collins [EMAIL PROTECTED]:
So you need to create a second call leg that is somewhat independent of the
first leg, so that you can play a file, and *then* bridge the new leg to the
current leg?
Yes, this is exactly what I want.
A normal bridge is nice, because it seems, that
to continue using socket outbound? Or will I perhaps find out, that I
can not do some things with socket inbound, which I could do with
socket outbound?
In the moment I just do not know which way to go...
Dennis
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wrong way.
Because many hours spent on freeswitch let me learn a lot about it and
let me know, that I can reach my goals with it, makes me not
resignate.
I hope I will find out, how to make both (or perhaps a lot more)
processes talk to each other.
Kind regards
Dennis
little tests to
understand more.
I have to admit, that I have to think a little bit more about what I
read, if it is written in english :-)
Thanks again
Dennis
2008/10/22 Anthony Minessale [EMAIL PROTECTED]:
You have to learn and understand the terminology and structure.
FSAPI is the name
and I never used IRC. Strange,
that I never used it, but I will be happy to take a ride. Might be
fun, a new experiance and informative.
Thanks
Dennis
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2008/10/22 Anthony Minessale [EMAIL PROTECTED]:
event socket has the command sendmsg which lets you send a message to a
specific channel. This can be any message but the one you are familiar with
is the one that tells the session to execute an application. Think of it as
you are sending an
of the
caller/inbound in p #1)
call-command: execute
execute-app-name: playback
execute-app-arg: /var/www/freeswitch/again_hello.wav
...the file will still be played back for the target/outbound.
Thanks again for your patience
Dennis
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2008/10/23 Anthony Minessale [EMAIL PROTECTED]:
What does 123 lead to? launching another script?
The 123 is the destination_number in the /dialplan/default.xml.
extension name=test
condition field=destination_number expression=^123$
action application=set
2008/10/23 Michael Collins [EMAIL PROTECTED]:
What is your IRC nick? Mine is mercutioviz. I'm interested in this issue
because I've been dialing in some somewhat similar scenarios and I might
be able to help, at least a little bit.
My nick is Dennis93. Looking forward to it.
Anybody else, who has an idea, what I could do (till I find someone in the IRC)?
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the way I need it?
Be sure I will be on IRC tomorrow. Perhaps this helps to avoid
missunderstandings ;-)
Thanks
Dennis
2008/10/26 Anthony Minessale [EMAIL PROTECTED]:
question what args do you pass to the socket app when you call it in
your dialplan.
are you using the full and async
have a one to many relationship with event socket to channels it can
control with 1 socket.
On Sun, Oct 26, 2008 at 1:37 PM, Dennis [EMAIL PROTECTED] wrote:
This is what I have in my /dialplan/default.xml:
extension name=test
condition field=destination_number expression=^123
.
If I am registered to events all, I a lot of information (to much),
but not with myevents. Any idea what I could do?
Thanks again for your great help
Dennis
2008/10/26 Anthony Minessale [EMAIL PROTECTED]:
try latest trunk, i think i can fix you issue by allowing sendmsg to work on
outside uuid
they are still there, at least if you register to myevents or all
events. i use the default settings from fs and get plenty of them.
2008/10/28 Andy Spitzer [EMAIL PROTECTED]:
Woof!
I used to get lots of variable_* lines when using socket_outbound. They
have disappeared. Is there something
with dtmf,
heartbeat and so on.
thanks
dennis
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hi,
i am using socket outbound and want to bridge two calls with each other.
i do the following:
a call comes in and it gets answered (inbound call). after i send the
answer to the inbound, i do an originate to park (outbound).
after i answered the outbound, i want both sides to be able to talk
channel_execute intercept and
channel_execute_complete intercept for call-direction outbound. but we
can not talk to each other and won't get any more information about
the status.
do you have a clue, what the problem could be or what we could try else?
regards,
dennis
2008/11/5 Birgit Arkesteijn [EMAIL
to talk to.
2.) if we make two originates, so that we have 2 outbound legs, we are
able to uuid-bride the two outbound legs with each other. it simply
does not work with the inbound...
thanks
dennis
2008/11/6 Anthony Minessale [EMAIL PROTECTED]:
If that doesn't work one of your uuids are wrong
an addition:
if we make a call to our socket (inbound), we get the following error
in our log:
2008-11-07 16:58:57 [ERR] switch_ivr.c:498 switch_ivr_park() Cannot
park channels that are under control already.
if an inbound comes in, we send a connect, then a park and then an
answer - nothing
and the outbound,
whatever we want. we can play different soundfiles to the different
uuid's, we can hangup calls with a specific uuid and so on.
thanks for you patience
dennis
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we read every single reply and we make socket_read with the legth
returned by Content-Length.
what we can do with fs, socket outbound and our php-script:
1.) we can answer the inbound, make a bridge to another phone and both
lines are connected and can talk to each other.
2.) we can make an
, while each freeswitch server can
go on handling call independantly.
but, would this setup cause any problems in case of stability or speech quality?
i have no idea, what task i should keep away from the application servers.
thanks,
dennis
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...
how can i move the inbound call or originated calls into a conference room?
it seems, as if one can do a lot with api in the conference room. but
i wonder what i have to do in the beginning, to put people (specific
uuid's) into a conference room.
thanks,
dennis
hi,
i would like to be able to listen to conversations, while they are
ongoing. this should not happen over a phone. i would like to be able
to have a link or something in my admin-area, where i can click, if i
want to listen to a conversation.
i thought about to start a record with socket
hi,
i have big problems with disconnects, when bridging and unbridging calls.
because i had random diconnects when testing fs from one softphone to
the other, i set up a little dialplan (socket outbound), to do some
hardcore testing.
after the inbound is answered, i do an originate to an
hi,
i wonder, if or what i do not understand how to do send_dtmf in the right way.
for example i want to send dtmf tones to my mobile mailbox the enter
the menu and do some changes to the settings. but whatever i try, it
does not work.
i tried:
sendmsg send_dtmf outbound_uuid 123#
sendmsg
so i would have to make a call with a phone to a specific dialplan? if
so, this would not be, what i whished (although it is nice to have the
option).
isn't there something, which can stream the voice of a given uuid? so
i could place a link in the html admin-area to spy an uuid and to hear
with the outbound.
if you want something different, please explain me a little more.
dennis
2008/11/28 Simon Tang [EMAIL PROTECTED]:
Hello,
I'm using event socket outbound, and have an issue where, after a bridge
ends and is terminated by Leg B, Leg A is also terminated. Here's the call
flow
we configured mod_shout and are able to record mp3. but if we start to
playback the file, it will only be played back to that point, which
was recorded, when we started the player.
we do this with api uuid_record uuid start /var/www/test.mp3.
we are also able to playback a (radio-)stream to an
i am using the latest svn trunk from today.
2008/12/2 Brian West [EMAIL PROTECTED]:
Are you on SVN trunk or what rev are you trying to use?
/b
On Dec 2, 2008, at 7:48 AM, Dennis wrote:
it seems, that fs has to stream to recording file to a streaming
server (like icecast), right
, at 9:03 AM, Dennis wrote:
i am using the latest svn trunk from today.
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sorry, problem solved :-)
it works very good with icecast2.
2008/12/2 Brian West [EMAIL PROTECTED]:
And you have your shoutcast/icecast server set up and functional?
/b
On Dec 2, 2008, at 9:03 AM, Dennis wrote:
i am using the latest svn trunk from today
is nothing one should use)?
and so on, and so on
i would be very happy to hear some user experiences with fs and fax.
if it seems, that we can use fax with over socket outbound, we will do
hardcore testing ;-)
thanks,
dennis
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a DCN while waiting for a DIS
fax_result_text = The HDLC carrier did not stop in a timely manner
fax_result_text = Unexpected message received
could someone please tell us, where the problem might be?
thanks
dennis
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, which we send for testing (not over fs or the same machine) are
sent over isdn.
2008/12/4 Michael Collins [EMAIL PROTECTED]:
Dennis,
Thanks for your input on the fax stuff! We will check this out and report
back.
Question: if one of the devs would like to SSH into your system to do
further
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 loglevel
with sendmsg playback send: loops: -1
2008/12/6 Faisal Maqsoodi [EMAIL PROTECTED]:
Hi,
Is there any built-in function, like playback, which plays a file again
and again unless interrupted. I want to use a simple function not FIFO.
Faisal
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nice built in features.
there are lots of dialplan samples delivered with fs and the wiki will
help you to start with the rest.
dennis
2008/12/6 Faisal Maqsoodi [EMAIL PROTECTED]:
I need some more help. I used send msg this way. Is there anything missing
bcoz its not working. Plz let me know what
not exist.
perhaps this is a feature, but i think that it would be nicer and more
reliable, if the sendmsg is only executed on the given uuid. if the
given uuid does not exist, nothing should happen or even nicer, an
event with an error should be sent to the socket.
thanks
dennis
changes to the default code?
thanks
dennis
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Huh?
src/switch_core_session.c vom line 899 to 901:
if (seconds 10) {
seconds = 60;
}
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#2 was because when you sendmsg with no uuid on an outbound socket it
defaults to the session who called you.
I changed to code to make a distinction between not supplying a uuid and
supplying an invalid uuid.
anthony, thanks for the quick reaction!
we just tested you changes and it works
great, that works! thanks a lot!
just tested the changes according an error, when a file is missing.
thanks again!
2008/12/8 Anthony Minessale [EMAIL PROTECTED]:
done
On Mon, Dec 8, 2008 at 9:18 AM, Dennis [EMAIL PROTECTED] wrote:
Huh?
src/switch_core_session.c vom line 899 to 901
i have to shift places. will be back in a few minutes and test.
no, we are using the simple sendmsg uuid hangup. as far as we
remember, we do not use api uuid_kill, because we do not get a hangup
event with this.
2008/12/8 Anthony Minessale [EMAIL PROTECTED]:
try the sendmsg issue again
are
you would get a hangup event in either case.
yes, you are right. we just tested and saw that. the reason for
sendmsg hangup, was the sometimes useful event-lock.
it works with api uuid_kill as we wanted. but with sendmsg hangup it
still does not work. shouldn't sendmsg hangup work like
going on on fs1.
now we have to redirect the call from fs2 to fs1. is this done with
redirect and some according settings/params or are there other ways
to do this? we would like to do this without our carrier doing
something, to be a little more independant.
thanks
dennis
, 2008, at 10:36 AM, Dennis wrote:
i would like to know, what the best way is, to redirect an incoming
call from one fs (fs1) to another fs (fs2).
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and his possibilities.
2008/12/18 Brian West br...@freeswitch.org:
What switch is your provider using?
/b
On Dec 18, 2008, at 10:52 AM, Dennis wrote:
i had a look at the deflect app, but as far as i understand it, the
carrier has to support/understand it ans react on the signals
it for good reason.
/b
On Dec 18, 2008, at 11:07 AM, Dennis wrote:
is deflect, what i understand? the provider has to support it? if yes,
what could i tell and ask the provider, to find a solution to this
problem? the provider is quite open for new ideas, although we do not
want
sorry, this is to difficult for me. what does that mean?
they pass a call to one of our fs. then we see, that the call should
be on another fs. we know, that the call is on the wrong fs, before we
send an answer. so we could react accordingly.
2008/12/18 Brian West br...@freeswitch.org:
do
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to redirect to.
Gabe
Dennis wrote:
i would like to know, what the best way is, to redirect an incoming
call from one fs (fs1) to another fs (fs2).
we use 3 freeswitch
thanks for all your help!
this sounds interesting. it seems, that these codes should be
available by default with sip!? is this right?
i will talk to the carrier tomorrow and ask, what is possible.
as far as i can see, i am always dependant on the carrier? there is no
way to pass a call from
sendmsg redirect to an ip-adress of one of our fs server works great.
thanks for your help.
dannis
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it's me again about mod fax... it is short before christmas and my
whish is, to get mod fax working quite reliable. is this possible
under optimal conditions?
all our tests lead by far to more failed faxes than received faxes. i
really like the fax feature and would like to see it beeing usable.
christmas (till i contact you because of some consulting
for final checks ;-)
dennis
2008/12/19 Anthony Minessale anthony.miness...@gmail.com:
You don't know where the audio goes after that switch in the same room until
it gets to the guy
with the fax machine.
No it will not be improved
dennis
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to a point where it levels off for that load... As
the loa decreases memory is not released but used for later when loading
increases again
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 23 Dec 2008 09:52:43 +0100
To: Freeswitch-users
of the server?
our setup is a new xeon quad core, 4 gb ram and ubuntu 64-bit. we also
entered the ulimit lines and set manage-presence to false.
thanks
dennis
2008/12/23 Ken Rice kr...@suspicious.org:
Freeswitch can handle a large volume of call... I suggest you review your
configs to make sure you
because the latest result was with the 9998, it can't be out app (at
the moment).
so there are no other typical things or settings i could look for?
2008/12/23 Ken Rice kr...@suspicious.org:
There are a number of issues you can be running into... It really depends on
how your app works, what
the 9998 is an extension in the default.xml to test with media flowing
through the line.
2008/12/23 Ken Rice kr...@suspicious.org:
Whats this 9998 to which you refer?
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 23 Dec 2008 10:43:30 +0100
sorry, i do not really understand what you mean with: Try the echo
tester but be sure you are using the media refector with sipp or you
arent doing anything useful.
what is the echo tester and what is media refector and how could i use it?
i would like to find out, how many people can talk to
stream
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 23 Dec 2008 11:06:06 +0100
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
sorry, i do not really understand what you mean
is it possible to define a profile and its params for a conference
dynamically over socket outbound?
in the moment, if we want to have multiple profiles for different
clients, we (have to) setup a profile in the conference.conf -
otherwise we get an error in fs.
because we have multipple
other functions).
is there a way, to NOT let the other side hear the dtmf sound (but of
course still make fs listening to it)?
thanks for the help
dennis
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event, because the socket was not
started. therefore i will not know, if the target is busy (hangup,
hangup cause: user busy).
it would be very helpful, if the socket would start immediately on an
event like channel originate.
thanks for the help
dennis
?
the basic problem for us, that, if we just want to make dialouts, we
are missing the inbound call to start the socket.
kind regards
dennis
2009/2/9 Anthony Minessale anthony.miness...@gmail.com:
when an originate is unsuccessful the failure and the cause code is returned
as the reply
that is interesting. we are receiving the dtmf digits over 2833. might
it be possible, that we receive 2833 AND inband (we asked our carrier
for 2833, because we had problems with inband and fs - and we got it)?
is there something we can setup in fs or is it a problem wich only our
carrier can
and beat them with a cluebat.
/b
On Feb 11, 2009, at 10:42 AM, Dennis wrote:
that is interesting. we are receiving the dtmf digits over 2833. might
it be possible, that we receive 2833 AND inband (we asked our carrier
for 2833, because we had problems with inband and fs - and we got
, Dennis wrote:
i can't tell, if they are sending both, but it seems so. we get 2833
for sure. they were kind enough to give it to us, because inband seems
to be quite unreliable over sip.
how can in find out, if both are coming and is there a way to block
inband to test?
perhaps we need both
switch_ivr_park() Cannot park channels that have no read codec.).
2009/2/10 Dennis oderm...@googlemail.com:
yes, you are right. we are receiving the reply.
but, we are using socket outbound and manage all calls over this
socket. we also measure the durations (like variable_duration and
variable_billsec
this does not help. we are using socket outbound and everything worked
before the changes yesterday.
we have the same error with other dialplans.
2009/2/11 Brian West br...@freeswitch.org:
Try answer or pre_answer before park.
/b
On Feb 11, 2009, at 12:37 PM, Dennis wrote:
anthony, did
regarding our problem.
2009/2/11 Brian West br...@freeswitch.org:
Please collect the backtrace and report it on Jira.
/b
On Feb 11, 2009, at 2:11 PM, Dennis wrote:
this does not help. we are using socket outbound and everything worked
before the changes yesterday.
we have the same error
ip-adresses?
kind regards
dennis
2009/5/13 Antonio Gallo ga...@mctelefonia.com:
Dennis ha scritto:
does someone know callweaver and can tell me, if there are some
important settings to be set for making it work with fs in the middle?
Look at this, i needed to apply it using a Patton gateway
to see all entries
(inbound/outbound) of one call.
is this possible?
kind regards
dennis
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that it was
nearly impossible to hear the other side talk, while the soundfile was
playing.
we tried uuid_displace uuid start /path/to/soundfile/soundfile.wav 0
mux 0.3, so that the loudness of soundfile only would be 30% - but
this does not work.
thanks kind regards
dennis
!?
anyway, if there is no other/better way, we have to do it with sox.
no, we are not using stereo-files.
kind regards
dennis
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dennis
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-phones and with them we have
the problems.
the big question is: wo or what might cause/trigger the hangup? is it
freeswitch or something else? we have a firewall (IPCop) - might there
be a setting, which needs to be set, to avoid theses problems?
kind regards
dennis
the second phonecall, the led of p2 is on, but the led of p1
is off, although p1 is still talking.
is there something we can do about this?
thanks kind regards
dennis
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ahh, I understand the issue now. Please open a jira on jira.freeswitch.org
for this issue.
ok, we could not imagine, that this behavior is meant to be.
we will try to open a jira with this issue (never opened a jira before).
kind regards
dennis
?
is it possible, that snom does not support a REAL session timer?
sonus is not involved.
kind regards
dennis
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sorry, but i do not know i which category i have to set this problem.
could you help me with that?
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Is the snom firmware up to the latest?
yes, the firmware is the latest.
I believe session timers should work properly with snom?
i don't know, but i think so. we tested with session timers off and
with setting the session timer to 0. both does not change anything.
we played with using a
Are the phones behind the same nat as FS?
no, both phones are behind the same nat, fs is behind another nat (and
the internet is inbetween).
both phones are in our office and we are sitting behind a nat. we
connect through our nat over the internet into the other nat, where fs
is behind.
are you setting presence_id=u...@domain variable on the outbound leg?
This is done for you in the DP via the user/ channel in the defaults but if
you are not using this
you have to set it manually.
in directory default we have the following:
params
param name=dial-string
this is the line (without stun - so we only have one leg) and we
called the 5900 to moh:
2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup
sofia/internal/1...@212.18.215.102 [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
we called the 5900 and waited 2 minutes...
or did you mean something
the pastebin number is 10129
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I think the problem is the session-timeout is too long and your nat mapping
is being deleted.
we changed the param session-timeout (in internal) to 20 and 40,
without success.
we changed the minimum-session-expires to 20, although we knew, the
allowed minimum is 90 and 90 was shown in the
If you look at your trace the call is sending a re-invite over and over and
over again with no reply
you need to examine your network topology and find out why the packets FS is
sending to your phone
never make it.
also try disabling session-timers on the snom
are you talking about our
-by-digit, but what about the fs-side?
thanks and kind regards
dennis
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the thing we want to make working nicer is the following:
we want the main/basic phonenumber (123456) to be reachable, so that
the telephone rings. but we also want it to be expandable with
ddi-digits.
example: dial the 123456 to reach the company, dial the 123456 1 to
reach the support.
in the
once you have 123456 won't you still be unsure if he will type the next 1 or
not and be forced to refuse it and wait anyway?
basically you are right. BUT, we know, that a basic phone number has 6
digits - so, we do not have to check anything before. as soon as we
have 6 digits, we look in our
ok, we will try this with the cirpack of our carrier. this will take
some days, till everything is set up.
after the tests i will come back to report.
2009/10/15 Anthony Minessale anthony.miness...@gmail.com:
right you can reply 484 in your dp at any time
action application=respond data=484
ok, as written, i come back after some tests with fs and a thomson cirpack.
it did not work - at least in our tests.
we are using socket outbound and when a call comes in, it starts the
socket of fs. the number may be 123456. fs sends the respond 484 and
our carrier receives this information.
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