[asterisk-users] Problem with Read() ?

W, I have also tested NOT using the Local/NXXNXX dialing and actually using a Dial to our upstream carrier, still with the U option, and same results. So, I honestly don't think it is an issue with using Local channels. Any thoughts? Am I just blatantly missing something? Or is somethin

Re: [Asterisk-Users] DID on analog line

ing in a T1 and setup a channel bank for you. I've always wondered why some places charge so much for DIDs, though. Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084

Re: [Asterisk-Users] Busy not jumping n + 101 anymore

) command, and create s-BUSY, s-CONGESTION, etc. in the same context.) Once I get around to updating our dialplans, that's what I plan on doing. Someone please correct me if I am wrong. *dons asbestos armor, just in case* Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]>

Re: [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6

: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free

Re: [Asterisk-Users] Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500

y after updating the config and rebooting. Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771

Re: [Asterisk-Users] kill a .call file

___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD

[Asterisk-Users] Round-robin with Queue

P/112, 2 for SIP/102, etc.) That just resulted in only SIP/100 being rung. So, what am I missing/doing wrong here? :) Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473

Re: [Asterisk-Users] Caller Name: Asterisk reading too fast

s.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/Sky

Re: [Asterisk-Users] All Page ??

I could dispose of the beep and just tell people to wait a few seconds, and that would work, and would probably work for you. Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.4

Re: [Asterisk-Users] Overriding Caller ID

any suggestions? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --

Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain

e greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED<[EMAIL

Re: [Asterisk-Users] Polycom SoundPoint 501 power adapter

it: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 url: http://www.winworld.cc/ __

Re: [Asterisk-Users] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone

gesting some diagnostic steps and/or solutions. Cheers - Stephen Joyce ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/

[Asterisk-Users] Weird issues with TDM400P

or resource busy I think this may have something to do with getting a dialtone instead of reorder after hangup (the first thing I mentioned.) Not 100% sure though. Anyone have any ideas on any of these? If you can share I'd appreciate it. TIA. Jeremy -- Jeremy Gault&

Re: [Asterisk-Users] Re: call does not hangup after client quits

Steve, If I am understanding your situation correctly (i.e. you are using a SIP client and then forcibly disconnecting/shutting it off during a call) you may want to look at your sip.conf for a setting called rtptimeout. This may do exactly what you want. When on a SIP call, and you disconne