Do any of the Obi products support DTMS dial tones? For example, when calling an AT&T bridgeline, you may need to enter a passcode followed by # to join a call.
Quote from: osprey92 on February 13, 2013, 06:15:13 AM
Do any of the Obi products support DTMS dial tones? For example, when calling an AT&T bridgeline, you may need to enter a passcode followed by # to join a call.
Yes the OBi adapters support various methods of sending dtmf over in-progress calls:
DTMF Method:
RFC2833
InBand
SipInfo
Auto
They also have a setting of yes/no
UseFixedDurationRFC2833DTMF
Hi hwittenb!
So, what do you do? Enter through the dial pad? Program a speed dial? Do these methods work? Are there alternative methods that I haven't mentioned?
Quote from: hwittenb on February 13, 2013, 08:19:41 AM
Quote from: osprey92 on February 13, 2013, 06:15:13 AM
Do any of the Obi products support DTMS dial tones? For example, when calling an AT&T bridgeline, you may need to enter a passcode followed by # to join a call.
Yes the OBi adapters support various methods of sending dtmf over in-progress calls:
DTMF Method:
RFC2833
InBand
SipInfo
Auto
They also have a setting of yes/no
UseFixedDurationRFC2833DTMF
Quote from: Phillip on February 13, 2013, 10:50:19 AM
So, what do you do? Enter through the dial pad? Program a speed dial? Do these methods work? Are there alternative methods that I haven't mentioned?
You have to enter the dtmf tones thru your dial pad. Speed dials and pauses, etc. do not work once the call has been initially connected.
Generally setting up the dtmf transmit method of RFC2833 works the best and most voip providers support and recommend that method of sending the tones "out of band". As you know once a call is initially connected a digital stream of rtp packets is sent to the distant location. "InBand" means that the tones are sent just as part of the normal voice stream. For "InBand" to work the analog to digital conversions must be very precise. "Out-of-Band" means that specific discrete packets for the tones are sent in addition to the rtp voice stream packets. RFC2833 is a standard protocol devised to solve problems with sending the dtmf tones in-band. Sip Info was another out of band protocol that preceded rfc2833. The Cisco/Linksys ata adapters called rfc2833 AVT.
If you have trouble sending dtmf tones over voip, with test calls, you typically try all the different settings and stick with the one that works with your voip provider.
hwittenb, I read your post nodding sagely as I went and actually understood what was said by the time I got to the end (a rarer occurrence than I would like to admit :)). This is another great piece of information that I will bookmark and save in a special set of folders I made for this purpose. Your efforts are appreciated. Thank you very much.