I am up and running with FreePBX and a 2 sip trunks and I have a couple of issues.
One way speech on outgoing/incoming SIP calls. The caller can hear me but I can’t hear them - all they here is themselves. Extn to Extn calls are fine!
Suggestions as to where to look?
I have set 3 digit extns and I would like to configure the system so it dials the exten as soon as there is an extension match rather then having to press “call” every time. I realize this might conflict with local numbers but I am willing to risk it. Is this possible
Oh is anyone using the “Skype add in” and what do you think! Waa it easy to configure?
Sorry for the lack of details.
One Way Speech.
Yes I did search for answers but didn’t find anything specific to my issues. I will search for NAT traveral options - thanks for the suggestions.
I am using the Polycom IP 550 phones - I did read the admin guide but missed any reference to that configuration.
Yes thats it Skype for Askerisk.
Next time I will add the critical information. Sorry for being a dense Tadpole.
One of the most common Asterisk issues, caused by NAT traversal options not set right, did you search for answers?
Since you did not tell us what kind of phones you are running it is impossible to give you an answer. Whatever phone you have you must not have read the administrators guide or you would have seen the “dial plan” or “digit match” options.
Since you did not tell us what kind of system you are running it is impossible to comment on this.
I also don’t know what a “skype add in” is. Digium sells a “Skype for Asterisk” channel driver, is that what you are talking about?