Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? -- jay austad | 612.423.1433 | austad@signal15.com
Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote:
Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second.
Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down?
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | austad@signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote:
Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running?
/b
On May 6, 2009, at 1:44 PM, Jay Austad wrote:
Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second.
Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down?
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Hi Jay, Have to say my DTMF works flawlessly on thousands of calls. (SVN trunk from a couple of days ago. We handle around 100,000 calls/day via FS) That said, I've found it depends on your SIP trunk provider. That doesn't mean to say there isn't a problem; it's just that I haven't come across it. Know it's not helpful, but there you go. Regards, ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jay Austad Sent: 06 May 2009 19:57 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition flaky I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | austad@signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote: Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com <http://www.cluecon.com/> _______________________________________________ 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
Hi Jay, Did you make a wireshark trace yet? You should be able to find out exactly what's going on there, which protocol is used, etc. We've had our share of problems with DTMF over SIP trunks as well. Your problems could also be related to timing issues introduced by multiple gateways. Do you know some details on voicepulse's network? There's lots of variations in implementation out there, unfortunately not always fully compatible. Good luck, Remko Van: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Namens Jay Austad Verzonden: woensdag 6 mei 2009 20:57 Aan: freeswitch-users@lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] DTMF recognition flaky I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | austad@signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote: Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com <http://www.cluecon.com/> _______________________________________________ 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
Remko Kloosterman <R.Kloosterman@mtel.nl> wrote:
Did you make a wireshark trace yet? You should be able to find out exactly what's going on there, which protocol is used, etc. We've had our share of problems with DTMF over SIP trunks as well.
I've just discovered that I'm having a similar problem to the one discussed in this thread. Here are the symptoms. 1. If I call FreeSWITCH from my Snom 320 SIP phone, DTMF recognition works perfectly. This is also true if I call a friend's FreeSWITCH system. 2. If I call a certain VoIP provider from the Snom phone, via FreeSWITCH (phone -> FreeSWITCH -> provider) and call the provider's DTMF test, DTMF recognition fails to work. Apparently this provider accepts only RFC-2833, which is what FreeSWITCH should be issuing - I haven't changed the settings in the external profile from the defaults. 3. If I call the same provider's DTMF test from PortAudio and issue the pa dtmf command, the provider recognizes the DTMF traffic correctly. I couldn't find any obvious configuration errors on the phone or in my internal and external Sofia profiles. I'll gladly run tshark if that's the next step to take. I can also try setting <param name="pass-rfc2833" value="true"/> in the internal profile, but this shouldn't be necessary, since as the wiki states in documenting this variable, FreeSWITCH should decode and re-encode the RFC2833 data anyway when this is set to false. I'll keep working on this, but in the meantime, suggestions are welcome.
you may have a sonus infection try some of the stuff from here under DTMF http://wiki.freeswitch.org/wiki/RTP_Issues On Thu, May 7, 2009 at 5:16 AM, Jason White <jason@jasonjgw.net> wrote:
Remko Kloosterman <R.Kloosterman@mtel.nl> wrote:
Did you make a wireshark trace yet? You should be able to find out exactly what's going on there, which protocol is used, etc. We've had our share of problems with DTMF over SIP trunks as well.
I've just discovered that I'm having a similar problem to the one discussed in this thread. Here are the symptoms.
1. If I call FreeSWITCH from my Snom 320 SIP phone, DTMF recognition works perfectly. This is also true if I call a friend's FreeSWITCH system.
2. If I call a certain VoIP provider from the Snom phone, via FreeSWITCH (phone -> FreeSWITCH -> provider) and call the provider's DTMF test, DTMF recognition fails to work. Apparently this provider accepts only RFC-2833, which is what FreeSWITCH should be issuing - I haven't changed the settings in the external profile from the defaults.
3. If I call the same provider's DTMF test from PortAudio and issue the pa dtmf command, the provider recognizes the DTMF traffic correctly.
I couldn't find any obvious configuration errors on the phone or in my internal and external Sofia profiles.
I'll gladly run tshark if that's the next step to take.
I can also try setting <param name="pass-rfc2833" value="true"/> in the internal profile, but this shouldn't be necessary, since as the wiki states in documenting this variable, FreeSWITCH should decode and re-encode the RFC2833 data anyway when this is set to false.
I'll keep working on this, but in the meantime, suggestions are welcome.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale <anthony.minessale@gmail.com> wrote:
you may have a sonus infection
try some of the stuff from here under DTMF
Thank you for the suggestion. I tried both the Sonus and Cisco settings in the external profile (running sofia profile external restart reloadxml after making the changes). This didn't help, unfortunately. If I were to make an informed guess, I would expect Cisco equipment to be at the other end, since my ISP has a strong relationship with Cisco. Whatever their solution for carriers is, this is likely to be it, but I could be wrong, of course. I find it interesting that dtmf over PortAudio works, but from the Snom phone it does not.
I've narrowed this problem down. When I call my ISP's DTMF test and issue DTMF from the Snom phone, do_2833() from switch_rtp.c is never called, as evidenced by freeswitch.log. However, if I call a friend's FreeSWITCH box from the phone (via my FreeSWITCH instance), do_2833() is called. It is also called if I use the voicemail extension on my local FreeSWITCH. Finally, if I call my ISP via PortAudio and use the pa dtmf command, do_2833() is called. It's either something in my configuration, or a bug. I'll keep looking. Anyone with ideas is welcome to offer suggestions.
Sorry for all the e-mail... If I turn off the jitter buffer that I had set in the dialplan extension for that provider, DTMF is correctly sent and detected by the other side. I suspect a bug, but maybe this is the desired behaviour.
Sound bugish to me - or at least not desired behavior. I'd suggest opening up a jira (jira.freeswitch.org) with as much documentation as you have so it can be researched and resolved. On Fri, May 8, 2009 at 3:46 AM, Jason White <jason@jasonjgw.net> wrote:
Sorry for all the e-mail...
If I turn off the jitter buffer that I had set in the dialplan extension for that provider, DTMF is correctly sent and detected by the other side.
I suspect a bug, but maybe this is the desired behaviour.
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-- -Rupa
Also, in general, I believe you want the jitter buffers on the end-point devices only. Not the guy in the middle. So, jitter buffer should be enabled on the phone, not within FS -- unless FS is the endpoint (eg: IVR). On Fri, May 8, 2009 at 7:05 AM, Rupa Schomaker <rupa@rupa.com> wrote:
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much documentation as you have so it can be researched and resolved.
On Fri, May 8, 2009 at 3:46 AM, Jason White <jason@jasonjgw.net> wrote:
Sorry for all the e-mail...
If I turn off the jitter buffer that I had set in the dialplan extension for that provider, DTMF is correctly sent and detected by the other side.
I suspect a bug, but maybe this is the desired behaviour.
_______________________________________________ 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
-- -Rupa
-- -Rupa
Rupa Schomaker <rupa@rupa.com> wrote:
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much documentation as you have so it can be researched and resolved.
If someone could add it to Jira, I'll detail the issue here. The Jira Web interface is a problem for me, and it doesn't seem to allow submissions by e-mail or in other ways. Basically, the problem is that RFC2833 DTMF isn't sent to the other side if a jitterbuffer is set in the dial plan extension for the outbound call with <action application="set" data="jitterbuffer_msec=180"/> and the call originates from my SIP phone (a Snom 320). The FreeSWITCH logs show that do_2833() in switch_rtp.c isn't called in this case. I'll gladly provide further details if and when anyone has a chance to investigate, assuming that it isn't desired behaviour (which in my opinion it isn't).
participants (7)
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Anthony Minessale
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Brian West
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Jason White
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Jay Austad
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Nik Middleton
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Remko Kloosterman
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Rupa Schomaker