I have realized a small Asterisk system with speech to text (Google speech Recognition ) and text to speech (Amazon Polly) capabilities, to help a small call center doing his job.
The System seats behind a NEC pbx, that pass call to Asterisk and then, call could go back.
It happens that on first IVR sometimes DTMF are not recognized.
I have even a voip number to do test and with this no problem.
Has anyone idea how and where to investigate or what to ask for to NEC’s dealer ?
I’m waiting for dealer to ask codec question…
Don’t know if usefull but, from NEC pbx call land to Asterisk as guest…and call go back to NEC via Dial(SIP/299@192.168.x.xxx)
That’s a pretty insecure chan_sip configuration. Unless there is a specific reason to use chan_sip, you should be using chan_pjsip.
There is really no point in obfuscating 192.168 addresses, as they are only valid within your LAN.
You will be getting your current settings from the general section of chan_sip, but the normal advice would be to include allowguest=no there and provide a specific section for the peer (except that the current advice would be to use chan_pjsip).
As you seem to be working blind and with no real knowledge of VoIP, I think it is going to be essential that you provide us a copy of what is going over the wire, and your complete sip.conf.