VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
austad at signal15.com
Guest
|
 Posted: Wed May 06, 2009 1:48 pm Post subject: [Freeswitch-users] DTMF recognition flaky |
 |
|
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
_______________________________________________
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 |
|
| Back to top |
|
 |
brian at freeswitch.org
Guest
|
| Posted: Wed May 06, 2009 1:49 pm Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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:
| Quote: | 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 (brian@freeswitch.org)
-- Meet us at ClueCon! http://www.cluecon.com |
|
| Back to top |
|
|
austad at signal15.com
Guest
|
| Posted: Wed May 06, 2009 2:02 pm Post subject: [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 (austad@signal15.com)
On May 6, 2009, at 1:46 PM, Brian West wrote:
|
|
| Back to top |
|
|
nik.middleton at noble...
Guest
|
| Posted: Wed May 06, 2009 3:43 pm Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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 (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 (brian@freeswitch.org)
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 |
|
| Back to top |
|
|
R.Kloosterman at mtel.nl
Guest
|
| Posted: Thu May 07, 2009 2:05 am Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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 (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 (brian@freeswitch.org)
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 |
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
| Posted: Thu May 07, 2009 5:17 am Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
Remko Kloosterman <R.Kloosterman@mtel.nl> wrote:
| Quote: |
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.
_______________________________________________
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 |
|
| Back to top |
|
|
anthony.minessale at g...
Guest
|
| Posted: Thu May 07, 2009 7:12 am Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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 (jason@jasonjgw.net)> wrote:
| Quote: | Remko Kloosterman <R.Kloosterman@mtel.nl (R.Kloosterman@mtel.nl)> wrote:
| Quote: |
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.
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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
|
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
| Posted: Thu May 07, 2009 7:19 pm Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
Anthony Minessale <anthony.minessale@gmail.com> wrote:
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.
_______________________________________________
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 |
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
| Posted: Fri May 08, 2009 2:19 am Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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.
_______________________________________________
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 |
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
|
| Back to top |
|
|
rupa at rupa.com
Guest
|
| Posted: Fri May 08, 2009 7:09 am Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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 (rupa@rupa.com)> wrote:
| Quote: | 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 (jason@jasonjgw.net)> wrote:
--
-Rupa
|
--
-Rupa |
|
| Back to top |
|
|
rupa at rupa.com
Guest
|
| Posted: Fri May 08, 2009 7:13 am Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
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 (jason@jasonjgw.net)> wrote:
--
-Rupa |
|
| Back to top |
|
|
jason at jasonjgw.net
Guest
|
| Posted: Fri May 08, 2009 8:40 pm Post subject: [Freeswitch-users] DTMF recognition flaky |
|
|
Rupa Schomaker <rupa@rupa.com> wrote:
| Quote: | 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).
_______________________________________________
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 |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
