Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=rfc2833 canreinvite=no Any ideas? Am I doing something wrong? Thanks! -JD-
We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband and make sure you're using a ulaw connection. If you use a lossy codec, it will scramble the DTMF tones. Your config would change like so, [sipphone] type=peer host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=inband canreinvite=no disallow=all allow=ulaw -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 10:27 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] DTMF issues with SIPPhone? Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=rfc2833 canreinvite=no Any ideas? Am I doing something wrong? Thanks! -JD- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Yes we are. I just double checked our line, and oddly, the dtmf tones aren't getting sent to our asterisk server. Switched it back to rfc2833, and it works. It was the other way around when I first connected us. Some informal testing just now doesn't show the DTMF tone problem in rfc2833 mode that you're having. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone? Louie, On 8/8/05, Tarpo, Louie <louie.tarpo@adamaircraft.com> wrote:> We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband and make sure you're using a ulaw connection. If you use a lossy codec, it will scramble the DTMF tones.Are you using SIPPhone? When I use dtmfmode=inband, it just doesn't recognize the tones at all.. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RFC2833 is sent out of band. What's the output on your asterisk console? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone? So the way I understand this is with rfc2833, DTMF is sent out of band. So does this mean that SIPPhone is interpreting the tones incorrectly? Asterisk shouldn't be doing any actual tone detection with this method, right? On 8/8/05, Tarpo, Louie <louie.tarpo@adamaircraft.com> wrote:> Yes we are. I just double checked our line, and oddly, the dtmf tones aren't getting sent to our asterisk server. Switched it back to rfc2833, and it works. It was the other way around when I first connected us. Some informal testing just now doesn't show the DTMF tone problem in rfc2833 mode that you're having._______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I find a verbosity of 10 (asterisk -rvvvvvvvvvv) gets me adequate logging for my purposes. I've been really pounding on our sipphone number the past half hour or so and I'm seeing the same issues you are. Sometimes it hits correctly, sometimes it doesn't. IE, Dialing 5954, some of the times it works, sometimes it's 5594, 5994, 5944, 59, 95, 54, etc. I know I'm not fatfingering the dialing because my cell prints the dtmf digits to the screen. We haven't been seeing the issues here because our sipphone number isn't published (yet). Louie -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone? On 8/8/05, Tarpo, Louie <louie.tarpo@adamaircraft.com> wrote:> RFC2833 is sent out of band. What's the output on your asterisk console?I don't see any output during this time on my asterisk console. Unless there's additional logging I'd need to enable? Thanks for the help! -JD- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users