I’m testing an Asterisk system with a few Cisco 7960s (SIP), and I’m having a problem when trying to dial anything that has a dtmf menu when I’m on speaker phone, like entering a password for a conference call bridge, etc. These are all calls going out a POTS line via a Zap interface. DTMF works fine when using asterisk voice mail, or other local things.
After much experimentation, I’ve determined that the remote service I’m trying to use is actually picking up the sound of the DTMF from the Cisco phone via the speaker phone. If i push buttons while on mute – no problems, off mute, I get duplicate entries.
I’ve searched through SIPDefaults.conf to see if there is a setting that would affect this, and I can’t find anything. Has anyone else come across this?