Translating DTMF from RFC2833 to INFO
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:
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 <yehavi.bourvine@gmail.com>wrote:
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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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@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 < yehavi.bourvine@gmail.com> wrote:
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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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 Yehavi Bourvine wrote:
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@gmail.com <mailto:oseslija@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 <yehavi.bourvine@gmail.com <mailto:yehavi.bourvine@gmail.com>> wrote:
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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Hello Metik, 2009/12/6 Metik <freeswitch-users-list@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 correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):
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. Thanks, __Yehavi:
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@metik.com <mailto:freeswitch-users-list@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 correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):
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.
Thanks, __Yehavi: ------------------------------------------------------------------------
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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@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@metik.com
<mailto:freeswitch-users-list@metik.com>>
You previously stated that your Cisco gateway has some "bug" that > prevents you from us...
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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@gmail.com>
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@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@metik.com
<mailto:freeswitch-users-list@metik.com>>
You previously stated that your Cisco gateway has some "bug" that > prevents you from us...
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participants (4)
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Anthony Minessale
-
Metik
-
Ognjen Seslija
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Yehavi Bourvine