VoIP Mailing List Archives

[odermann at googlemail...]

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
digit-by-digit, but what about the fs-side?


thanks and kind regards
dennis

_______________________________________________
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 at g...]

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 <odermann@googlemail.com (odermann@googlemail.com)> wrote:

[odermann at googlemail...]

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
test it and how we can see if it is supported or not.

any hint?


2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>:

[tculjaga at gmail.com]

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 <odermann@googlemail.com (odermann@googlemail.com)> wrote:

[anthony.minessale at g...]

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 <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:

[freeswitch-users-list ...]

Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s. -metik ----- Original Message ----- From: Anthony Minessale (anthony.minessale@gmail.com) To: freeswitch-users@lists.freeswitch.org (freeswitch-users@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 <tculjaga@gmail.com (tculjaga@gmail.com)> wrote: 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 <odermann@googlemail.com (odermann@googlemail.com)> wrote: how could we try? we played arround with a snom phone (snom seems tosupport something in this direction, but are not shure, how we cantest it and how we can see if it is supported or not.any hint?2009/10/13 Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)>: > 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 <odermann@googlemail.com (odermann@googlemail.com)> wrote:>>>> 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>> digit-by-digit, but what about the fs-side?>>>>>> thanks and kind regards>> dennis>>>> _______________________________________________>> FreeSWITCH-users mailing list>> FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)>> MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])> IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])> googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])> pstn:213-799-1400>> _______________________________________________> FreeSWITCH-users mailing list> FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)> FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])pstn:213-799-1400 _______________________________________________FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org

[tculjaga at gmail.com]

i never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability.

BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck...

T.

On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list@metik.com (freeswitch-users-list@metik.com)> wrote:

[freeswitch-users-list ...]

As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over the years, I would avoid it if at all possible. What does it particularly do that can not accomplished by using RFC 2833 or (less ideal) inband DTMF? Or are you attempting to use it as a band-aid to address some sort of interop issue with the carrier involved that is wrecking havoc with your particular application? -metik ----- Original Message ----- From: Tihomir Culjaga (tculjaga@gmail.com) To: freeswitch-users@lists.freeswitch.org (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 issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability.BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck...T. On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list@metik.com (freeswitch-users-list@metik.com)> wrote: Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s. -metik ----- Original Message ----- From: Anthony Minessale (anthony.minessale@gmail.com) To: freeswitch-users@lists.freeswitch.org (freeswitch-users@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 <tculjaga@gmail.com (tculjaga@gmail.com)> wrote: 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 <odermann@googlemail.com (odermann@googlemail.com)> wrote: how could we try? we played arround with a snom phone (snom seems tosupport something in this direction, but are not shure, how we cantest it and how we can see if it is supported or not.any hint?2009/10/13 Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)>: > 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 <odermann@googlemail.com (odermann@googlemail.com)> wrote:>>>> 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>> digit-by-digit, but what about the fs-side?>>>>>> thanks and kind regards>> dennis>>>> _______________________________________________>> FreeSWITCH-users mailing list>> FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)>> MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])> IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])> googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])> pstn:213-799-1400>> _______________________________________________> FreeSWITCH-users mailing list> FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)> FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])pstn:213-799-1400 _______________________________________________FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _______________________________________________FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org

[tculjaga at gmail.com]

I suppose he want to have a central dialplan and a dummy phone instead... something as a MGCP phone behavior.

T.

On Tue, Oct 13, 2009 at 10:22 PM, Metik <freeswitch-users-list@metik.com (freeswitch-users-list@metik.com)> wrote:

[odermann at googlemail...]

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 moment our carrier waits x (milli-)seconds to see, if the
123456 was dialed or if there were more digits attached. after this
time, the carrier sends us the whole number as a block (may it be
123456 or 1234561) and we answer.

we want to avoid the waiting time the carrier needs to wait for
(possible) more digits. we want to receive every single digit from our
carrier and we want to tell fs, when the number is complete and fs can
answer.

_______________________________________________
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 at g...]

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 refuse it and wait anyway?

This feature is mostly used to allow the server to accept the call the instant it knows there is an exact match in the dialplan or a non-existent number has been dialed.  For instance, if 1234 was a unique ext, it could instantly return since it knows.


On Wed, Oct 14, 2009 at 9:31 AM, Dennis <odermann@googlemail.com (odermann@googlemail.com)> wrote:

[odermann at googlemail...]

[anthony.minessale at g...]

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 itself so test it and see.  I may need to dig up the answer again from the sofia dev.



On Thu, Oct 15, 2009 at 2:47 AM, Dennis <odermann@googlemail.com (odermann@googlemail.com)> wrote:

[odermann at googlemail...]

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.minessale@gmail.com>:

[odermann at googlemail...]

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 fs ends the call with
hangup_cause = invalid_number_format.
the carrier has one more digit for the phone number and sends 1234567
and the above mentioned behavior repeats.

the behavior we want and expected is, that the call stays in the
socket after response 484, so that the carrier can send the 1234567
into the same socket.
the management, when fs should send response 484 and when fs should be
answered would be programmed by us.
it also important, that fs keeps the call in the socket, so we can
tell fs, to answer the call after x seconds anyway.

any ideas, what we could do?

_______________________________________________
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