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
I'm going to say No!
/b
On Nov 4, 2009, at 2:23 AM, Dennis wrote:
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
You cannot.
This is how the sip spec works.
Every new invite is a new call and a new trip to the dialplan.
You will probably need to design your code to send the appropriate 484 and
be prepared to exit and be called again with the new digits.
On Wed, Nov 4, 2009 at 2:23 AM, Dennis
Brian is right,
pls, lets stop with exceptions and get stick to RFCs... otherwise it will be
a big mess ...
T.
On Wed, Nov 4, 2009 at 3:03 PM, Brian West br...@freeswitch.org wrote:
I'm going to say No!
/b
On Nov 4, 2009, at 2:23 AM, Dennis wrote:
is there a way to send something like
hi anthony,
i believe, that there is no problem with the communication between fs
and the cirpack (everything works to smooth as if this could be
possible). if fs sends the 484, the cirpack sends more digits to fs
(if there are some), so this works as it should. the problem is, that
fs ends the
The patch was it's ability to accept subsequent invites.
Your problem is that in sip each new attempt to send an invite is another
call.
484 is a final response so the call with too few digits is terminated.
On Tue, Nov 3, 2009 at 9:57 AM, Dennis oderm...@googlemail.com wrote:
hi anthony,
i
This may be possible with tcp, how could this work on udp? Can you
provide an rfc reference on this?
Mike
On Oct 24, 2009, at 8:13 AM, Dennis wrote:
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
On Oct 24, 2009, at 8:13 AM, Dennis wrote:
ok, as written, i come back after some tests with fs and a thomson
cirpack.
No idea if this is useful as I'm a noob with fs. If not please excuse
the noise. In the past Asterisk to work properly with Cirpack needed the
following patch:
diff -uNr
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.
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
right you can reply 484 in your dp at any time
action application=respond data=484 Address Incomplete/
then it should try again.
The bit i can't remember is if we committed a certain 1 line patch that
makes sofia parse the next invite to the same call properly, the patch was
to the sofia lib
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
?
-metik
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:* freeswitch-users@lists.freeswitch.org
*Sent:* Tuesday, October 13, 2009 3:24 PM
*Subject:* Re: [Freeswitch-users] SIP Overlap support?
i never found it working properly... i always had some interoperability
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
So with overlap you will have to keep refusing the call until the right
amount of digits are dialed.
This mode would send 1 then 12 then 123 then 1234 then 12345 then 123456 as
they were being dialed.
once you have 123456 won't you still be unsure if he will type the next 1 or
not and be forced to
hi there,
i would like to ask, if fs has support for something like SIP Overlap?
instead of receiving the phonenumber from our carrier in a block, we
want to receive the phonenumber digit-by-digit and we want to tell fs
when the number is complete. our carrier could send us the phonenumber
have you tried it?
I *think* either we did support it 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 oderm...@googlemail.com wrote:
hi there,
i would like to ask, if fs has support for something like SIP
you need a softswitch i'm afraid a SIP phone is not designed for
overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM, Dennis oderm...@googlemail.com wrote:
how could we try? we played arround with a snom phone (snom seems to
support something in this direction, but are not shure, how we can
i do think some softphone can do it but i forgot which one it was either
snom or grandstream
On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga tculj...@gmail.comwrote:
you need a softswitch i'm afraid a SIP phone is not designed for
overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM,
@lists.freeswitch.org
Sent: Tuesday, October 13, 2009 2:01 PM
Subject: Re: [Freeswitch-users] SIP Overlap support?
i do think some softphone can do it but i forgot which one it was either snom
or grandstream
On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga tculj...@gmail.com wrote:
you
have a device
somewhere in the middle that doesn't support 484s.
-metik
- Original Message -
*From:* Anthony Minessale anthony.miness...@gmail.com
*To:* freeswitch-users@lists.freeswitch.org
*Sent:* Tuesday, October 13, 2009 2:01 PM
*Subject:* Re: [Freeswitch-users] SIP Overlap support
-users] SIP Overlap support?
i do think some softphone can do it but i forgot which one it was either
snom or grandstream
On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga tculj...@gmail.com
wrote:
you need a softswitch i'm afraid a SIP phone is not designed for
overlap
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