Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like:
[8315551212] type => friend ... dtmfmode => inband ...
When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the other end? What I'm noticing is that if I call a pstn line using an entry like this through asterisk, and then press digits on the SIP phone connected to asterisk, I hear very short tones on the pstn line instead of the long tones I generate on the SIP phone. In addition, if I press digits too quickly on the SIP phone, where "too quickly" is not very fast at all, many digits are dropped entirely and do not make it to the pstn phone at all. It occurred to me that this might be a fixable problem in the Asterisk source code, but when I read the code itself, it is not clear to me who is generating these short dtmf bursts, and perhaps it is the fault of the SIP instrument, a Cisco 7960 running SIP image 6.2, it self. So, if anyone can explain to me where the DTMF tones are coming from when the dtmfmode is set to "inband", I'd be most appreciative.
sip.conf does not use =>, it uses =. i.e. type=friend
With inband dtmf the device that you press the key on will generate the DTMF. If the codec is ulaw or alaw the DTMF will come thru loud and clear. If the codec is not ulaw or alaw then the DTMF will be corrupted by the audio compression and you will hear a very short sound or a garbled tone. Of course if the phone is set for RFC2833 and asterisk is set for inband then you may not hear anything at all.
--Eric
--
I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org.
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