Hi, i have a OpenVox TDM400p and when i dial to pstn, 6 seconds before i get the ring tone of the PSTN,
then i edited the string from toneduration=100 to toneduration=40, now i have a delay of 4 seconds.
But i want to know if exist a string to edit somewhere to make this delay more short. *like 1 secs. 
were you able to get a fix for this problem? please send me an email if you were able to fix this . Also if anyone out there knows how to fix this, please get back to me at tdivr@yahoo.com. I am running asterisk 1.8 and openvox AE2410P and there is about a 6 to 8 second delay before the calls go out the pstn, so the calls come in to the asterisk server SIP and after about 6 or 8 seconds it goes out the openvox card.
I need someone to configure asterisk for wholesale,I need just the basic asterisk to accept a call SIP and send it out over openvox AE2410P FXO cards. I need someone to set this up correctly. I have this set up working now butthe problemi have is there is a 6 to 8 sec delay from the time the call is answered by asterisk and when it start to go out over the FXO card. I need to elimate this delay. I also need to make sure there are no audio files or ivr playing on the gateway. I need it configured so if the ports on the asterisk terminating gateway is full, it woul immediately send back a 503 to the customer. please email me if you can get this done. Letme know how many hours it would take to do this and how much you want to do it $) email me at tdivr@yahoo.com
You’ve tail-ended an unrelated thread with your request for consultancy.
There is a jobs forum for such requests.
Regarding the other question, this typically happens if the incoming side allows overlap dialing and the Asterisk dialplan permits extra digits. If the origination is from SIP phones, it could be a problem with the dialplan (different type) in the phone itself.