SETUP: Asterisk@Home 0.8 install (Asterisk 1.0.7, AMP v1.10.007, plus misc) new RHEL3 certified Dell Poweredge server, new Cisco 7960G’s, new TDM400P w/4fxo’s, four SBC Centrex lines config’d for direct dial (no ‘9’ etc.).
ISSUE: Outbound dialing to PSTN is intermittently erratic. That is, I dial a local or long distance number and sometimes it connects to the correct number and other times it calls the wrong number. Some 877 numbers return the message “You must dial a 1 to…” message consistently even though there is a 1. However an analog handset connected to the PSTN line directly dials all #'s perfectly.
DEBUG: Asterisk log shows phone number being correctly sent from Cisco SIP phone to Asterisk, and the correct number being submitted to Dial.
Help? Any ideas? Our inbound is great, however outbound looking not so good.
Solution to my own request (compliments of a gentleman on the Asterisk@Home forums):
Add “w+” or “ww+” (0.5 seconds per “w”) to the Outbound Dial Prefix for your Zaptel Trunk in the AMP interface (assuming you are using it or an Asterisk@Home version that includes it).
Why the Zaptel needs a delay before dialing to correctly dial your numbers is a mystery, however, this solution does work! Curious if any Asterisk savvy developers might have an asnwer to this curious issue (e.g. does it do it on all machines, only faster machines, or what)?
It is probably not a problem with your card so much as you need to give the telephone exchange time to setup when going off hook. Humans usually wait to hear a dialtone. Perhaps Digium could put in a delay before dialing, or maybe there is one configurable for the card already. Maybe they can be set to do dialtone detection like a modem.
I am just pondering as I don’t use internal cards.