Local extensions not recognized on outbound call

When making an outbound call from local sip/zap -> asterisk -> remote number ie. 021234567, any buttons I press while I’m on the phone will be recognized by remote side only, no effect to local extensions. Is there way to let asterisk recognize local extensions during remote call?

not sure exactly what you mean

if i am understanding you right,
you have a sip phone, which connects to asterisk, and then when you dial digits into the sip phone and the other end gets them, ie for voice mail system, bank remote access, customer service, etc.
you want the call to come into Asterisk, and whoever called can then get a voice menu and make a selection by dialing
right?

Assuming i am answering the right question,
how or if this works depends on how * is connected to the telephone network (PSTN). If you are using a zaptel card, it should work automatically without any tweaking (IE zaptel card and PSTN line from your telco). If you are using a SIP provider things are a bit different.
There are three ways to send DTMF over a SIP call. First you must understand a SIP call has two parts- the control signalling, which is the actual SIP protocol, going on port 5060 (this is just a text based setup protocol). Once the call starts then the actual audio data (media) goes over RTP on another port, asterisk defaults to 10000-20000 range.
The first is inband. This sends the DTMF as its audio tones inside the audio stream. This only works if you are using the ulaw or alaw codec, other codecs distort the tones. Inband is also usually the least reliable.
Second is RFC2833. With RFC, the DTMF tones are stripped out of the audio and sent along with the RTP data as events. This means that instead of transmitting the sound of you pressing 1, the sound is stripped from the audio stream and the remote end is told “the user pushed 1”. This is generally much more reliable.
Lastly is SIP INFO. It’s very similar to RFC, however the data is sent via the SIP control channel as opposed to the RTP media channel.

For DTMF to work right with *, you have to figure out what your provider is sending you. Chances are it’s either inband or RFC2833, INFO isn’t often used. in sip.conf you should play with the dtmfmode= setting for the SIP provider you use. Also try the ‘auto’ setting which will choose between inband and RFC based on what is available.

Hope that helped

What are trying to do?

Only certain key strokes will be recognised. see features.conf for them

Ian

i am going on the assumption that he wants to set up an IVR of some kind and he wants asterisk to recognize incoming dtmf…