is there a way to send something like 484 (or something else), which
does not make it a final answer and keep the call/socket alive?
so we can ask the cirpack for further digits and decide what to do, if
the cirpack does not send any digits.
2009/11/3 Anthony Minessale :
> The patch was it's ab
84 without ending the session/socket
and waiting for an answer of the cirpack? we would take care of the
rest.
kind regards,
dennis
2009/10/15 Anthony Minessale :
> right you can reply 484 in your dp at any time
>
>
> then it should try again.
>
> The bit i can't rememb
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. but
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 :
> right you can reply 484 in your dp at any time
>
>
> then it should try again.
>
> The bit i can't remember is
> 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
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
or we would with a small patch to sofia
> lib that I cannot recall if we ever got committed.
>
>
> On Tue, Oct 13, 2009 at 8:51 AM, Dennis wrote:
>>
>> hi there,
>>
>> i would like to ask, if fs has support for something like "SIP Overlap"?
>>
>
honenumber
digit-by-digit, but what about the fs-side?
thanks and kind regards
dennis
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> 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
> 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 con
the pastebin number is 10129
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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 differe
> 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:
everthing works fine with the led l
> 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.
_
> 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 st
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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s
disabling works?
is it possible, that snom does not support a REAL session timer?
sonus is not involved.
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
bout the leds.
kind regards
dennis
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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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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
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dennis
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!?
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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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
see all entries
(inbound/outbound) of one call.
is this possible?
kind regards
dennis
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o two ip-adresses?
kind regards
dennis
2009/5/13 Antonio Gallo :
> 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
des
communicate with eachother?
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?
thanks & kind regards
dennis
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regarding our problem.
2009/2/11 Brian West :
> 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.
>
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 :
> Try answer or pre_answer before park.
>
> /b
>
> On Feb 11, 2009, at 12:37 PM, Dennis wrote:
>
>
_ivr.c:674
switch_ivr_park() Cannot park channels that have no read codec.).
2009/2/10 Dennis :
> 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
> variab
e the carrier just sends them as normal sound, which is
played as a tone, without beeing used for dtmf?
2009/2/11 Brian West :
> turn on the start_dtmf app and dial digits from the outside.. if you
> get duplicate digits then they are sending both.
>
> /b
>
> On Feb 11, 2009, at
at them 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
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 sol
originates?
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 :
> when an originate is unsuccessful the failure and the cause code is returned
> as the reply to the originate r
ones for dtmf.
> 2) post a bounty to have FS clip the audio for x milliseconds when a tone is
> detected. (you will still hear faint clicks between the start of the tone
> and when the clipping activates)
>
>
>
> On Mon, Feb 9, 2009 at 8:59 AM, Dennis wrote:
>>
>>
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
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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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 fs-server
1/2 the media stream
>
>
>> From: Dennis
>> Reply-To:
>> Date: Tue, 23 Dec 2008 11:06:06 +0100
>> To:
>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>
>> sorry, i do not really understand what you mean with: "Try the ech
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
the 9998 is an extension in the default.xml to test with media flowing
through the line.
2008/12/23 Ken Rice :
> Whats this 9998 to which you refer?
>
>
>> From: Dennis
>> Reply-To:
>> Date: Tue, 23 Dec 2008 10:43:30 +0100
>> To:
>> Subject: Re: [Freesw
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 :
> There are a number of issues you can be running into... It really depends on
> how your app works, what your actual configu
eview to get more out 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 :
> Freeswitch can handle a large volume of call... I suggest you review your
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
>> Reply-To:
>> Date: Tue, 23 Dec 2008 09:52:43 +0100
>> To:
>> Subject: [Freeswitch-users] Memory que
?
thanks
dennis
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hi,
i am quite new to freeswitch and now i finally have fs up and running
as i want with my php scripts to handle the calls.
now that i want to start the service in the near future, i would like
to test the performance of the whole system and the reliability.
what are your experiences and ideas,
ably be much more reliable once it can do T38.
>
> Be happy with what you have for the holiday season.
>
>
>
> On Fri, Dec 19, 2008 at 10:44 AM, Dennis wrote:
>>
>> it's me again about mod fax... it is short before christmas and my
>> whish is, to ge
christmas (till i contact you because of some consulting
for final checks ;-)
dennis
2008/12/19 Anthony Minessale :
> 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 by Chri
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.
sendmsg redirect to an ip-adress of one of our fs server works great.
thanks for your help.
dannis
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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 one
lure
> 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 (
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 :
> do they follow a 302 redi
son.
>
> /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
and his possibilities.
2008/12/18 Brian West :
> 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 th
18, 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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rence 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.
rmal_clearing
>
> in place of
> call-command: execute
> execute-app-name: hangup
> execute-app-arg: normal_clearing
>
>
> On Mon, Dec 8, 2008 at 10:56 AM, Dennis <[EMAIL PROTECTED]> wrote:
>>
>> > you would get a hangup event in either case.
>>
>&g
> 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 uuid_kil
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
>
>
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?
>
> #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
> Huh?
src/switch_core_session.c vom line 899 to 901:
if (seconds < 10) {
seconds = 60;
}
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this
change.
might it be possible, to do the same changes to the default code?
thanks
dennis
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it, if the outbound_uuid does 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
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
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
>
>
> __
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
, 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 SS
ved 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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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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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 t
/b
>
> On Dec 2, 2008, at 9:03 AM, Dennis wrote:
>
>> i am using the latest svn trunk from today.
>
>
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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 str
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 u
d helps nothing 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 terminate
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
everyt
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 playb
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 outboun
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 inboun
i am using socket outbound and if an inbound call comes in, i answer
the call and play a soundfile for the caller.
if the caller presses the dtmf key * the playback is stopped and i
receive an channel_execute_complete playback for this file. this only
happens with the * - the other keys do nothing
, which asks to enter the pin...
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 con
d fail, 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,
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 inbo
you are right, the shown script is very simple. we shortened it a lot
just to show the problem.
in these tests we do not need any events, because we just answer and
originate - for these actions we do not need any events, filters and
so on.
the rest we do in the cli. in the cli we do show channels
the php process!?
thanks
dennis
2008/11/7 Anthony Minessale <[EMAIL PROTECTED]>:
> Can you capture the whole log from the instant you get the inbound call
> until you give up on the uuid bridge?
> I don't see any of the log about your outbound call.
>
> are you doin
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 els
ith the inbound 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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riginated calls hangs up, the other call is also beeing hangup.
we just can't see, what we are doing wrong or what we could try or
change to make it work.
thanks,
dennis
2008/11/6 Anthony Minessale <[EMAIL PROTECTED]>:
> you are missing execute-app-arg
>
> sendmsg
> cal
w tested over our carrier, so that there are only two uuid's. the
result is the same with only two possible uuid's (see log in
http://pastebin.freeswitch.org/6019).
i really have no clue, what i am doing wrong.
thanks a lot
dennis
2008/11/6 Anthony Minessale <[EMAIL PROTECTED]>
nt 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 on
he
inbound), we get 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 B
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
with dtmf,
heartbeat and so on.
thanks
dennis
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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 some
en play a soundfile or make an uuid_bridge.
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 y
e you the
> uuid of the new call if it's successful and you can
> then send it messages just like the other one.
>
> you can 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 <[E
ctions, but they do not work
for me (at least I do not know how to use it). Are there any settings
I have to configure to make it work 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
Anybody else, who has an idea, what I could do (till I find someone in the IRC)?
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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.
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