Linksys SPA 3000 As I understand it (and I could be wrong) the S0 at the end of the dial plan means send the previous digits without wait for time-out. But something like [xx.S0] will not work as intended because of the "." which means one or more digits
Here is my dial plan: (<#:*97>|[*23]xxS0|[469]xxxxxxxxxS0|1[2-9]xxxxxxxxxS0|011xx.) Here's why: <#:*97> Replaces diallling *97 (voicemail) with "#" [*23]xxS0 Dials feature Codes and any extensions in the 200 and 300 range without waiting for timeout. [469]xxxxxxxxxS0 Dials any local number without waiting for timeout. 1[2-9]xxxxxxxxxS0 Dails any Long Distance number without waiting for timeout 011xx. Dials any International number (12 or 13 digits) RTP Packet Size: should be set to 0.020 for Asterisk Hope this helps Henry L.Coleman [VoIP-PBX.ca] ------------------------------------------------- > Simon P. Ditner< > The "S0" means end of input, so what you've effectively done is > require 3 or more digits of input before routing the call to Asterisk > by changing it to "xx.". If you set that to <S0:s>, it will try to > reach extension 's' at the default SIP proxy, or if you entered > <S0:123> it would try '123' > > On the "PSTN Line" screen there is an "Answer Delay" under the section > "FXO Timer Values (sec)". The default value is something like 16 > seconds. North American analog ringing is 2s on, 4s off, 2s on, 4s > off, etc.., and in my tests I found that it consistently captured > caller ID down as low as 5 seconds in that field. You should then only > be hearing 2-3 rings when calling in before asterisk picks up. > > To get logging from the device, you need to fill in the IP address of > a syslog server (i.e. your asterisk server) on the System tab. This > means that you need to have syslog configured to log messages from > remote devices. > > Hope that helps! > > Cheers, > spd > > 2009/8/16 Frank Bax <[email protected]>: >> Jim Van Meggelen wrote: >>> >>> Extension 's' will only work if the call arrives without an extension. >>> Your ATA is passing that extension. >> >> >> I did the initial setup of this device in April/May; and then could not get >> back to it until now. I searched through the SPA3102 config pages; under >> "Admin - Advanced - PSTN line", I found a section called "Dialplans". >> >> Dialplans 1-7 have "(xx.)"; whereas 8 has "S0<:[email protected]>". >> >> Thinking the 8th position had something to do with 5 rings; I exchanged >> values in position 1 & 8. Now asterisk answers after 3rd ring; but nothing >> appears in console with until I enter any extension number. >> >> >>> When you call, make sure you have the asterisk console up, with verbosity >>> set to 5. You'll be able to see when the signal to asterisk arrives. >> >> With verbosity set to 5; nothing appears on console until after 5th ring. >> After changing Dialplan settings on SPA3102 as above; I hear dialtone after >> 3 rings, but nothing appears on console until after I enter an extension >> number. >> >> >> --------------------------------------------------------------------- >> To unsubscribe, e-mail: [email protected] >> For additional commands, e-mail: [email protected] >> >> > > > > -- > | It ain't what you don't know that gets you into trouble. It's what > | you know for sure that just ain't so. -- Mark Twain > | > | Network: http://www.linkedin.com/in/spditner > | http://facebook.com/people/Simon-P-Ditner/776370031 > | http://twitter.com/spditner > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > > > --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
