[Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Yehavi Bourvine]

Hello,

I have Polycom phones which send only RFC-2833 (or inband which I dislike)
and they should go out to the PSTN via a Cisco gateway. The Cisco gateway
has some bug and accepts only INFO.

I did a few tests:

- Some of the phones are on different profile than the Cisco. On their
profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
'dtmf-type=info' and Freeswitch did the translation. All works ok...
- Some of the phones are on the same profile as the Cisco, so I must set
dtmf-type to rfc2833; it works with internal applications (like voicemail)
but does not work through the Cisco as it misinterprets the rfc2833


Is there a way to set some variable (or a parameter to the bridge
application) to do the translation?

Thanks! __Yehavi:
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[Ognjen Seslija]

Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param "dtmf-type" is valid only for outbound calls from the
profile.

Ognjen

On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine

Post by Yehavi Bourvine
Hello,
I have Polycom phones which send only RFC-2833 (or inband which I
dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco
gateway has some bug and accepts only INFO.
- Some of the phones are on different profile than the Cisco. On their
profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
'dtmf-type=info' and Freeswitch did the translation. All works ok...
- Some of the phones are on the same profile as the Cisco, so I must
set dtmf-type to rfc2833; it works with internal applications (like
voicemail) but does not work through the Cisco as it misinterprets the
rfc2833
Is there a way to set some variable (or a parameter to the bridge
application) to do the translation?
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[Yehavi Bourvine]

Hello Ognjen,

From the tests I've done it is not so... When I set the profile to use
INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the
FreeSwich ignores it (does not have phone-events field in the reply SDP)
which causes the phone to not send RFC2833 events...

Regards, __Yehavi:

2009/12/3 Ognjen Seslija <oseslija at gmail.com>

Post by Ognjen Seslija
Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param "dtmf-type" is valid only for outbound calls from the
profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine <

Post by Yehavi Bourvine
Hello,
I have Polycom phones which send only RFC-2833 (or inband which I
dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco
gateway has some bug and accepts only INFO.
- Some of the phones are on different profile than the Cisco. On their
profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
'dtmf-type=info' and Freeswitch did the translation. All works ok...
- Some of the phones are on the same profile as the Cisco, so I must
set dtmf-type to rfc2833; it works with internal applications (like
voicemail) but does not work through the Cisco as it misinterprets the
rfc2833
Is there a way to set some variable (or a parameter to the bridge
application) to do the translation?
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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[Metik]

You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?

Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" setting that does not agree
to well with FS's (default) "rfc2833-pt" setting, you should not have to
use (SIP) INFO unless you want to.

I would recommend doing the following to ensure you are hitting the
correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):

command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)

output:
DTMF Relay = enabled,
dtmf-relay = rtp-nte,

example:

show dialplan number 5551212 | i (dtmf-relay|DTMF Relay)
DTMF Relay = enabled,
dtmf-relay = rtp-nte,

Also, you can sift through "show sip-ua calls" for the call and ensure
that the value of "Negotiated Dtmf-relay" is "rtp-nte".

-metik

Post by Yehavi Bourvine
Hello Ognjen,
From the tests I've done it is not so... When I set the profile to
use INFO, and a phone calls and asks for RFC2833 (phone-events in the
SDP) the FreeSwich ignores it (does not have phone-events field in the
reply SDP) which causes the phone to not send RFC2833 events...
2009/12/3 Ognjen Seslija <oseslija at gmail.com <mailto:oseslija at gmail.com>>
Bear in mind that FS will accept both 2833 and INFO in any profile
on an inbound call. Param "dtmf-type" is valid only for outbound
calls from the profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine
Hello,
I have Polycom phones which send only RFC-2833 (or inband
which I dislike) and they should go out to the PSTN via a
Cisco gateway. The Cisco gateway has some bug and accepts only
INFO.
*
Some of the phones are on different profile than the
Cisco. On their profile I set 'dtmf-type=rfc2833' and on
the Cisco's profile I set 'dtmf-type=info' and
Freeswitch did the translation. All works ok...
*
Some of the phones are on the same profile as the Cisco,
so I must set dtmf-type to rfc2833; it works with
internal applications (like voicemail) but does not work
through the Cisco as it misinterprets the rfc2833
Is there a way to set some variable (or a parameter to the
bridge application) to do the translation?
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
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[Yehavi Bourvine]

Hello Metik,



2009/12/6 Metik <freeswitch-users-list at metik.com>

Post by Metik
You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?

It is a PSTN dialpeer here, and it cannot be defined on it...

Post by Metik
Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" setting that does not agree
to well with FS's (default) "rfc2833-pt" setting, you should not have to
use (SIP) INFO unless you want to.
I would recommend doing the following to ensure you are hitting the
command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)

Unfortunately this does not work on PSTN dial peers.

Post by Metik
Also, you can sift through "show sip-ua calls" for the call and ensure
that the value of "Negotiated Dtmf-relay" is "rtp-nte".

This indeed shows that it has negotiated rtp-nte. Even when I do debug for
CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them
while it accepts them via INFO. As I said: I guess this is a bug.

Since the gateway is on a remote site I hesitate on upgrading it until I hae
the chance to go there.

Thanks, __Yehavi:
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[Metik]

Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild. Unlike some other SIP feature servers, I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.

-metik

Post by Yehavi Bourvine
Hello Metik,
2009/12/6 Metik <freeswitch-users-list at metik.com
<mailto:freeswitch-users-list at metik.com>>
You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?
It is a PSTN dialpeer here, and it cannot be defined on it...
Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" setting that does not agree
to well with FS's (default) "rfc2833-pt" setting, you should not have to
use (SIP) INFO unless you want to.
I would recommend doing the following to ensure you are hitting the
command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)
Unfortunately this does not work on PSTN dial peers.
Also, you can sift through "show sip-ua calls" for the call and ensure
that the value of "Negotiated Dtmf-relay" is "rtp-nte".
This indeed shows that it has negotiated rtp-nte. Even when I do debug
for CCAPI events (I think) I see it decodes the DTMFs; however, it
ignores them while it accepts them via INFO. As I said: I guess this
is a bug.
Since the gateway is on a remote site I hesitate on upgrading it until
I hae the chance to go there.
------------------------------------------------------------------------
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[Anthony Minessale]

Some more bad news for you, info dtmf spec has expired and has been
abandoned. Wait till you see what they did accept instead......

On Dec 6, 2009 1:22 PM, "Metik" <freeswitch-users-list at metik.com> wrote:

Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild. Unlike some other SIP feature servers, I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.

-metik

Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik <
freeswitch-users-list at metik.com

Post by Yehavi Bourvine
<mailto:freeswitch-users-list at metik.com>>

You previously stated that your Cisco gateway has some "bug" that >

prevents you from us...

Post by Yehavi Bourvine
------------------------------------------------------------------------ >
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[Yehavi Bourvine]

Hello all,

*debug voip rtp session named-event*s shows that it receives and
understands the DTMFs, but it does not send them to the PSTN (sends only
those received via INFO). I haveto find some time and go to the remote site
to update to the latest IOS... I will update after this has been done.

Regards, __Yehavi:

2009/12/6 Anthony Minessale <anthony.minessale at gmail.com>

Post by Anthony Minessale
Some more bad news for you, info dtmf spec has expired and has been
abandoned. Wait till you see what they did accept instead......
Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".
Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.
I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild. Unlike some other SIP feature servers, I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.
Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.
-metik
Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik <
freeswitch-users-list at metik.com

Post by Yehavi Bourvine
<mailto:freeswitch-users-list at metik.com>>

You previously stated that your Cisco gateway has some "bug" that >

prevents you from us...

Post by Yehavi Bourvine
------------------------------------------------------------------------

_____________________...

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[Yehavi Bourvine]

Hello all,

*debug voip rtp session named-event*s shows that it receives and
understands the DTMFs, but it does not send them to the PSTN (sends only
those received via INFO). I haveto find some time and go to the remote site
to update to the latest IOS... I will update after this has been done.

Regards, __Yehavi:

2009/12/6 Anthony Minessale <anthony.minessale at gmail.com>

Post by Anthony Minessale
Some more bad news for you, info dtmf spec has expired and has been
abandoned. Wait till you see what they did accept instead......
Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".
Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.
I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild. Unlike some other SIP feature servers, I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.
Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.
-metik
Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik <
freeswitch-users-list at metik.com

Post by Yehavi Bourvine
<mailto:freeswitch-users-list at metik.com>>

You previously stated that your Cisco gateway has some "bug" that >

prevents you from us...

Post by Yehavi Bourvine
------------------------------------------------------------------------

_____________________...

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[Anthony Minessale]

Some more bad news for you, info dtmf spec has expired and has been
abandoned. Wait till you see what they did accept instead......

On Dec 6, 2009 1:22 PM, "Metik" <freeswitch-users-list at metik.com> wrote:

Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild. Unlike some other SIP feature servers, I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.

-metik

Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik <
freeswitch-users-list at metik.com

Post by Yehavi Bourvine
<mailto:freeswitch-users-list at metik.com>>

You previously stated that your Cisco gateway has some "bug" that >

prevents you from us...

Post by Yehavi Bourvine
------------------------------------------------------------------------ >
_____________________...

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[Metik]

Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild. Unlike some other SIP feature servers, I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.

-metik

Post by Yehavi Bourvine
Hello Metik,
2009/12/6 Metik <freeswitch-users-list at metik.com
<mailto:freeswitch-users-list at metik.com>>
You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?
It is a PSTN dialpeer here, and it cannot be defined on it...
Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" setting that does not agree
to well with FS's (default) "rfc2833-pt" setting, you should not have to
use (SIP) INFO unless you want to.
I would recommend doing the following to ensure you are hitting the
command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)
Unfortunately this does not work on PSTN dial peers.
Also, you can sift through "show sip-ua calls" for the call and ensure
that the value of "Negotiated Dtmf-relay" is "rtp-nte".
This indeed shows that it has negotiated rtp-nte. Even when I do debug
for CCAPI events (I think) I see it decodes the DTMFs; however, it
ignores them while it accepts them via INFO. As I said: I guess this
is a bug.
Since the gateway is on a remote site I hesitate on upgrading it until
I hae the chance to go there.
------------------------------------------------------------------------
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Yehavi Bourvine]

Hello Metik,



2009/12/6 Metik <freeswitch-users-list at metik.com>

Post by Metik
You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?

It is a PSTN dialpeer here, and it cannot be defined on it...

Post by Metik
Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" setting that does not agree
to well with FS's (default) "rfc2833-pt" setting, you should not have to
use (SIP) INFO unless you want to.
I would recommend doing the following to ensure you are hitting the
command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)

Unfortunately this does not work on PSTN dial peers.

Post by Metik
Also, you can sift through "show sip-ua calls" for the call and ensure
that the value of "Negotiated Dtmf-relay" is "rtp-nte".

This indeed shows that it has negotiated rtp-nte. Even when I do debug for
CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them
while it accepts them via INFO. As I said: I guess this is a bug.

Since the gateway is on a remote site I hesitate on upgrading it until I hae
the chance to go there.

Thanks, __Yehavi:
-------------- next part --------------
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[Metik]

You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?

Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" setting that does not agree
to well with FS's (default) "rfc2833-pt" setting, you should not have to
use (SIP) INFO unless you want to.

I would recommend doing the following to ensure you are hitting the
correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):

command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)

output:
DTMF Relay = enabled,
dtmf-relay = rtp-nte,

example:

show dialplan number 5551212 | i (dtmf-relay|DTMF Relay)
DTMF Relay = enabled,
dtmf-relay = rtp-nte,

Also, you can sift through "show sip-ua calls" for the call and ensure
that the value of "Negotiated Dtmf-relay" is "rtp-nte".

-metik

Post by Yehavi Bourvine
Hello Ognjen,
From the tests I've done it is not so... When I set the profile to
use INFO, and a phone calls and asks for RFC2833 (phone-events in the
SDP) the FreeSwich ignores it (does not have phone-events field in the
reply SDP) which causes the phone to not send RFC2833 events...
2009/12/3 Ognjen Seslija <oseslija at gmail.com <mailto:oseslija at gmail.com>>
Bear in mind that FS will accept both 2833 and INFO in any profile
on an inbound call. Param "dtmf-type" is valid only for outbound
calls from the profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine
Hello,
I have Polycom phones which send only RFC-2833 (or inband
which I dislike) and they should go out to the PSTN via a
Cisco gateway. The Cisco gateway has some bug and accepts only
INFO.
*
Some of the phones are on different profile than the
Cisco. On their profile I set 'dtmf-type=rfc2833' and on
the Cisco's profile I set 'dtmf-type=info' and
Freeswitch did the translation. All works ok...
*
Some of the phones are on the same profile as the Cisco,
so I must set dtmf-type to rfc2833; it works with
internal applications (like voicemail) but does not work
through the Cisco as it misinterprets the rfc2833
Is there a way to set some variable (or a parameter to the
bridge application) to do the translation?
_______________________________________________
FreeSWITCH-users mailing list
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[Yehavi Bourvine]

Hello Ognjen,

From the tests I've done it is not so... When I set the profile to use
INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the
FreeSwich ignores it (does not have phone-events field in the reply SDP)
which causes the phone to not send RFC2833 events...

Regards, __Yehavi:

2009/12/3 Ognjen Seslija <oseslija at gmail.com>

Post by Ognjen Seslija
Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param "dtmf-type" is valid only for outbound calls from the
profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine <

Post by Yehavi Bourvine
Hello,
I have Polycom phones which send only RFC-2833 (or inband which I
dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco
gateway has some bug and accepts only INFO.
- Some of the phones are on different profile than the Cisco. On their
profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
'dtmf-type=info' and Freeswitch did the translation. All works ok...
- Some of the phones are on the same profile as the Cisco, so I must
set dtmf-type to rfc2833; it works with internal applications (like
voicemail) but does not work through the Cisco as it misinterprets the
rfc2833
Is there a way to set some variable (or a parameter to the bridge
application) to do the translation?
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[Yehavi Bourvine]

Hello,

I have Polycom phones which send only RFC-2833 (or inband which I dislike)
and they should go out to the PSTN via a Cisco gateway. The Cisco gateway
has some bug and accepts only INFO.

I did a few tests:

- Some of the phones are on different profile than the Cisco. On their
profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
'dtmf-type=info' and Freeswitch did the translation. All works ok...
- Some of the phones are on the same profile as the Cisco, so I must set
dtmf-type to rfc2833; it works with internal applications (like voicemail)
but does not work through the Cisco as it misinterprets the rfc2833


Is there a way to set some variable (or a parameter to the bridge
application) to do the translation?

Thanks! __Yehavi:
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[Ognjen Seslija]

Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param "dtmf-type" is valid only for outbound calls from the
profile.

Ognjen

On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine

Post by Yehavi Bourvine
Hello,
I have Polycom phones which send only RFC-2833 (or inband which I
dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco
gateway has some bug and accepts only INFO.
- Some of the phones are on different profile than the Cisco. On their
profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
'dtmf-type=info' and Freeswitch did the translation. All works ok...
- Some of the phones are on the same profile as the Cisco, so I must
set dtmf-type to rfc2833; it works with internal applications (like
voicemail) but does not work through the Cisco as it misinterprets the
rfc2833
Is there a way to set some variable (or a parameter to the bridge
application) to do the translation?
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