I am having a problem with SendDTMF() - 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(wwwww3wwwww2wwwwww1wwwww4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 work and failed to do number 2. Sometime all went through fine. dtmfmode=rfc2833 are set in the sip.conf file How do I debug to see what went wrong and how to fix? Asterisk 1.8.12.0 Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located in UK) VOIP Provider in UK. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120516/0d2d7391/attachment.htm>
Ing CIP. Alejandro Celi Mariátegui
2012-May-18 16:41 UTC
[asterisk-users] 50% of time SendDTMF failed
Did you try putting inband parameter in dtmfmode and dtmf of your sip.conf? Regards, -- Ing CIP. Alejandro Celi Mari?tegui <alex at linux.org.pe> http://cipher.pe/web/asterisk.html El mi?, 16-05-2012 a las 16:07 +0100, Shahid H escribi?:> I am having a problem with SendDTMF() - 50% of time it did > not succeed. > > > > I suspect it is not sending clear DTMF tones to the IVR. > > > For example: > > > SendDTMF(wwwww3wwwww2wwwwww1wwwww4) > > > Sometime digit 3 and 2 work, and failed to do digit 1. > Sometime digit 3 work and failed to do number 2. > Sometime all went through fine. > > > dtmfmode=rfc2833 are set in the sip.conf file > > > How do I debug to see what went wrong and how to fix? > > > Asterisk 1.8.12.0 > Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located > in UK) > VOIP Provider in UK. > > > Thanks > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120518/7fc963fc/attachment.htm>
I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected dynamically. Wanted to check with the community if this feature holds true on latest versions of Asterisk ? Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Fri, May 18, 2012 at 10:11 PM, Ing CIP. Alejandro Celi <alex at linux.org.pe> wrote:> ** > > Did you try putting inband parameter in dtmfmode and dtmf of your sip.conf? > > Regards, > > > -- > Ing CIP. Alejandro Celi Mari?tegui > <alex at linux.org.pe> > http://cipher.pe/web/asterisk.html > > > El mi?, 16-05-2012 a las 16:07 +0100, Shahid H escribi?: > > I am having a problem with SendDTMF() - 50% of time it did not succeed. > > > > I suspect it is not sending clear DTMF tones to the IVR. > > > > For example: > > > > SendDTMF(wwwww3wwwww2wwwwww1wwwww4) > > > > Sometime digit 3 and 2 work, and failed to do digit 1. > > Sometime digit 3 work and failed to do number 2. > > Sometime all went through fine. > > > > dtmfmode=rfc2833 are set in the sip.conf file > > > > How do I debug to see what went wrong and how to fix? > > > > Asterisk 1.8.12.0 > > Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located in > UK) > > VOIP Provider in UK. > > > > Thanks > > --_____________________________________________________________________-- Bandwidth and Colocation Provided by http://www.api-digital.com --New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello > asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use <http://lists.digium.com/mailman/listinfo/asterisk-users>rs > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120519/cfdd8665/attachment.htm>