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[Freeswitch-users] SIP Overlap support?

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mike at jerris.com
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Posted: Sat Oct 31, 2009 10:35 am Post subject: [Freeswitch-users] SIP Overlap support?

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:

Quote:

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?

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freeswitch-list at puz...
Guest

Posted: Sat Oct 31, 2009 10:59 am Post subject: [Freeswitch-users] SIP Overlap support?

Quote:

On Oct 24, 2009, at 8:13 AM, Dennis wrote:

Quote:

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 asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
--- asterisk-1.4.19.org/main/rtp.c 2007-10-08 22:06:33.000000000 +0200
+++ asterisk-1.4.19/main/rtp.c 2007-11-11 13:12:28.000000000 +0100
@@ -1383,6 +1383,7 @@
[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
+ [96] = {0, AST_RTP_DTMF},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

Maybe this helps.

Regards,
Patrick

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brian at freeswitch.org
Guest

Posted: Sat Oct 31, 2009 11:08 am Post subject: [Freeswitch-users] SIP Overlap support?

We have <param name="rfc2833-pt" value="101"/> on the profile and you
to set this.

So you can set it to 96 if needed. But you shouldn't have to do that
if they say 96 and we say 101 they should be listening on 101 from us
and we should be listening on 96 from them... thats why its called an
RTP map.

/b

On Oct 31, 2009, at 10:48 AM, Patrick List wrote:

Quote:

On Oct 24, 2009, at 8:13 AM, Dennis wrote:

Quote:

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 asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
--- asterisk-1.4.19.org/main/rtp.c 2007-10-08 22:06:33.000000000 +0200
+++ asterisk-1.4.19/main/rtp.c 2007-11-11 13:12:28.000000000 +0100
@@ -1383,6 +1383,7 @@
[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
+ [96] = {0, AST_RTP_DTMF},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

Maybe this helps.

Regards,
Patrick

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odermann at googlemail...
Guest

Posted: Tue Nov 03, 2009 11:23 am Post subject: [Freeswitch-users] SIP Overlap support?

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 session/socket after a 484, so that the cirpack sends the
following digits into another socket.

you wrote about a "1 line patch", which might not have been
implemented - at least it seems so.

is there a way to get someone of the sofia devs to fix this small
problem, so that fs sends the 484 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 <anthony.minessale@gmail.com>:

Quote:

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.

anthony.minessale at g...
Guest

Posted: Tue Nov 03, 2009 12:36 pm Post subject: [Freeswitch-users] SIP Overlap support?

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

Quote:

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odermann at googlemail...
Guest

Posted: Wed Nov 04, 2009 3:34 am Post subject: [Freeswitch-users] SIP Overlap support?

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 <anthony.minessale@gmail.com>:

Quote:

brian at freeswitch.org
Guest

Posted: Wed Nov 04, 2009 9:18 am Post subject: [Freeswitch-users] SIP Overlap support?

I'm going to say No!

/b

On Nov 4, 2009, at 2:23 AM, Dennis wrote:

Quote:

anthony.minessale at g...
Guest

Posted: Wed Nov 04, 2009 10:57 am Post subject: [Freeswitch-users] SIP Overlap support?

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

Quote:

tculjaga at gmail.com
Guest

Posted: Wed Nov 04, 2009 11:04 am Post subject: [Freeswitch-users] SIP Overlap support?

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 <brian@freeswitch.org (brian@freeswitch.org)> wrote:

Quote:

I'm going to say No!

/b

On Nov 4, 2009, at 2:23 AM, Dennis wrote:

Quote:

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