Would this also be an issue/resolution with Google Voice as well? Using Dingaling I am routing my calls to an IVR to intercept the call from going to Google Voice voicemail. However, the IVR won't accept DTMF. On Mon, May 23, 2011 at 8:24 AM, Michael Collins <msc@freeswitch.org> wrote:

Sidharth, A mobile phone will always send DTMFs in-band, so you need to be ready for that scenario. I recommend you add this to your dialplan for inbound calls: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Test various scenarios and make sure they work - call from Skype, from mobile phone, from a land line, etc. Let us know what happens. -MC On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya <sid.kshatriya@gmail.com> wrote:

...

Dear Friends,

I'm using a voip carrier Voicenetwork.ca (recommended by someone on this list). While I'm generally happy with their service there seems to be one fatal problem: My IVR does not recognize DTMF!

I have set <param name="dtmf-type" value="rfc2833"/> in both sip_profile/internal.xml and sip_profile/external.xml

The symptom of the problem is that making a call via skype will always make the IVR recognize the DTMF while using something like a mobile phone almost always won't!

I've tried in-band detection too. I'm making international calls into my IVR and the reliability of the in-band detection is not so good, possibly because of the quality of the call.

Can someone please help me?

Thanks,

Sidharth

-- Sidharth Kshatriya www.sidk.info

_______________________________________________ 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

_______________________________________________ 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

Thanks for your reply, Michael. I have tried doing a start dtmf but then I've noticed: 1. My skype key presses get unreliable... 2. The dtmf detection is patchy with inband -- works sometimes and doesn't work sometimes :-) Any settings to keep in mind apart just adding a start dtmf ? Thanks, Sidharth On Mon, May 23, 2011 at 8:54 PM, Michael Collins <msc@freeswitch.org> wrote:

Sidharth,

A mobile phone will always send DTMFs in-band, so you need to be ready for that scenario. I recommend you add this to your dialplan for inbound calls: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

Test various scenarios and make sure they work - call from Skype, from mobile phone, from a land line, etc. Let us know what happens.

-MC

On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya < sid.kshatriya@gmail.com> wrote:

...

Dear Friends,

I'm using a voip carrier Voicenetwork.ca (recommended by someone on this list). While I'm generally happy with their service there seems to be one fatal problem: My IVR does not recognize DTMF!

I have set <param name="dtmf-type" value="rfc2833"/> in both sip_profile/internal.xml and sip_profile/external.xml

The symptom of the problem is that making a call via skype will *always*make the IVR recognize the DTMF while using something like a mobile phone *almost always* won't!

I've tried in-band detection too. I'm making international calls into my IVR and the reliability of the in-band detection is not so good, possibly because of the quality of the call.

Can someone please help me?

Thanks,

Sidharth

-- Sidharth Kshatriya www.sidk.info

_______________________________________________ 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

_______________________________________________ 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

-- Sidharth Kshatriya www.sidk.info

Just my two cents, but we had similar problems with mobile dtmf, and found the best combination was to set INFO rather than RFC2833 and to always use start_dtmf . That also allowed the soft phone clients to operate properly as well, and if the client is configurable it is best to tell the soft phone what type of dtmf to use other wise we got the same symtoms as you described, namely multiple key presses when you only press it once Regards Bob On Tue, May 24, 2011 at 10:29 PM, Sidharth Kshatriya <sid.kshatriya@gmail.com> wrote:

Thanks for your reply, Michael. I have tried doing a start dtmf but then I've noticed: 1.  My skype key presses get unreliable... 2. The dtmf detection is patchy with inband -- works sometimes and doesn't work sometimes :-)

Any settings to keep in mind apart just adding a start dtmf ? Thanks, Sidharth

In your SIP profile eg internal.xml change the RFC2833 to info. eg <param name="dtmf-type" value="info"/> Then just call start_dtmf at the start of the call, like you said and it definitely helped my cause Bob On Wed, May 25, 2011 at 5:04 PM, Sidharth Kshatriya <sid.kshatriya@gmail.com> wrote:

Thanks for your tips Bob. So I have done the following 1. Used start_dtmf before my lua IVR script starts 2. Asked my VOIP carrier to send me INFO events Now I need to know what I need to do on my internal and external SIP profiles in terms of config (if anything, from the default config). Currently when I get a call on an external line I simply bridge it to my internal IVR number. Thanks, Sidharth

I guess what I mean is this: he needs to listen to what ACTUALLY comes down the line. There's no substitute for the human ear. If the tones coming down the line are not clean or otherwise are not recognizable then I would suggest that looking for a solution inside of FreeSWITCH itself may not be too helpful... That being said, if the DTMFs are coming in cleanly then there's definitely something hinky going on inside his system. -MC On Tue, May 24, 2011 at 12:08 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:

Michael,

I'm not really sure what you mean by this...

The DTMF method used between an endpoint and a SIP carrier is largely determined by the SIP carrier, not the carrier or device on the far end.

On the PSTN DTMF is always inband. If configured for out of band DTMF it's up to the various media gateways in use by VoIP providers to detect the inband DTMF, squelch it from the G711 input audio, and create out of band events.

Of course if you're configured for inband and using G711 it will be inband all the way through.

P.S. - Cell phones either use AMR or EVRC these days and I would guess they're using out of band DTMF too. I've seen some documentation claiming support for inband DTMF on either but that seems strange to me...

On Mon, May 23, 2011 at 10:24 AM, Michael Collins <msc@freeswitch.org> wrote:

...

Sidharth, A mobile phone will always send DTMFs in-band, so you need to be ready for that scenario. I recommend you add this to your dialplan for inbound calls: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Test various scenarios and make sure they work - call from Skype, from mobile phone, from a land line, etc. Let us know what happens. -MC On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya <sid.kshatriya@gmail.com> wrote:

...

Dear Friends,

I'm using a voip carrier Voicenetwork.ca (recommended by someone on this list). While I'm generally happy with their service there seems to be

one

...

fatal problem: My IVR does not recognize DTMF!

I have set <param name="dtmf-type" value="rfc2833"/> in both sip_profile/internal.xml and sip_profile/external.xml

The symptom of the problem is that making a call via skype will always make the IVR recognize the DTMF while using something like a mobile phone almost always won't!

I've tried in-band detection too. I'm making international calls into my IVR and the reliability of the in-band detection is not so good, possibly because of the quality of the call.

Can someone please help me?

Thanks,

Sidharth

-- Sidharth Kshatriya www.sidk.info

_______________________________________________ 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

_______________________________________________ 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

-- Kristian Kielhofner

_______________________________________________ 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

The tech support guys from my VOIP carrier had the following to say: "Have you told freeswitch to ANSWER the call prior to the the IVR coming into play as by default it would answer EARLY MEDIA thus allowing only 1 way audio and no DTMF." What does this mean? I answer the call within the lua script. On Wed, May 25, 2011 at 12:52 AM, Michael Collins <msc@freeswitch.org>wrote:

I guess what I mean is this: he needs to listen to what ACTUALLY comes down the line. There's no substitute for the human ear. If the tones coming down the line are not clean or otherwise are not recognizable then I would suggest that looking for a solution inside of FreeSWITCH itself may not be too helpful...

That being said, if the DTMFs are coming in cleanly then there's definitely something hinky going on inside his system.

-MC

On Tue, May 24, 2011 at 12:08 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:

...

Michael,

I'm not really sure what you mean by this...

The DTMF method used between an endpoint and a SIP carrier is largely determined by the SIP carrier, not the carrier or device on the far end.

On the PSTN DTMF is always inband. If configured for out of band DTMF it's up to the various media gateways in use by VoIP providers to detect the inband DTMF, squelch it from the G711 input audio, and create out of band events.

Of course if you're configured for inband and using G711 it will be inband all the way through.

P.S. - Cell phones either use AMR or EVRC these days and I would guess they're using out of band DTMF too. I've seen some documentation claiming support for inband DTMF on either but that seems strange to me...

On Mon, May 23, 2011 at 10:24 AM, Michael Collins <msc@freeswitch.org> wrote:

...

Sidharth, A mobile phone will always send DTMFs in-band, so you need to be ready for that scenario. I recommend you add this to your dialplan for inbound calls: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Test various scenarios and make sure they work - call from Skype, from mobile phone, from a land line, etc. Let us know what happens. -MC On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya <sid.kshatriya@gmail.com> wrote:

...

Dear Friends,

I'm using a voip carrier Voicenetwork.ca (recommended by someone on

this

...

list). While I'm generally happy with their service there seems to be one fatal problem: My IVR does not recognize DTMF!

I have set <param name="dtmf-type" value="rfc2833"/> in both sip_profile/internal.xml and sip_profile/external.xml

The symptom of the problem is that making a call via skype will always make the IVR recognize the DTMF while using something like a mobile phone almost always won't!

I've tried in-band detection too. I'm making international calls into my IVR and the reliability of the in-band detection is not so good, possibly because of the quality of the call.

Can someone please help me?

Thanks,

Sidharth

-- Sidharth Kshatriya www.sidk.info

_______________________________________________ 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

_______________________________________________ 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

-- Kristian Kielhofner

_______________________________________________ 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

_______________________________________________ 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

-- Sidharth Kshatriya www.sidk.info

On Tue, May 24, 2011 at 10:08 PM, Sidharth Kshatriya < sid.kshatriya@gmail.com> wrote:

The tech support guys from my VOIP carrier had the following to say:

"Have you told freeswitch to ANSWER the call prior to the the IVR coming into play as by default it would answer EARLY MEDIA thus allowing only 1 way audio and no DTMF."

What does this mean? I answer the call within the lua script.

If you have something like "session:answer()" in your script then you're good. You can also call the "answer" app prior to sending the call to the Lua script. If you do a SIP trace and you see FS sending back a 200 OK then you are telling the VoIP provider that you've answered the call. If you see a 180 or 183 w/SDP but no 200 OK then that would indicate you've only established early media and not actually answered the call. It sounds like the carrier does not send audio in early media, which basically makes sense. Look at the early media article on our wiki if you need to learn more about it... -MC