similar to: upgrading from 1.4.39 to 1.8.5

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155 Anybody

Which file in Asterisk have a setting for time zone? When asterisk record incoming call in Master.csv the time is 6hr. ahead. I'm on: Canada/Mountain zone -- Joseph

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8 my caller ID is not working WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c976040515 -- Joseph

How can I monitor asterisk if all lines are registered etc? I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc. Usually I just restart asterisk and it solves the problem. Is there an application that will email me if case any line looses registration with with asterisk? Or any better solution! -- Joseph

Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet? I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy. -- Joseph

How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/Username Host Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server (null) (D) 255.255.255.255 0 Unmonitored voip

In my dial plan I have these two lines: exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) How to add "if" statement to execute these line only after let say 5pm. To record conversation only after 5pm. -- Joseph

I'm testing GotoIf($["${CALLERID(num) but I'm missing something as it is not working: [office-open] exten => s,1,Wait(1) exten => s,2,Answer() ; for Caller ID is 471-5665, always signal congestion: exten => s,3,GotoIf($["${CALLERID(num)}" = "4715665"]?4:6) exten => s,4,Playtones(congestion) exten => s,5,Congestion(5) exten =>

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

When I type: asterisk -rx "core show channels" I usually get Channel Location State Application(Data) SIP/pstn-4444-000003 7807574622 at internal: Up Dial(SIP/77807574622 at pstn-9998 SIP/pstn-9998-000003 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph

I receive several calls from this scamer: Senior SafeAlert It is an automated call and they keep rotating their caller ID so it is harder to block them. Does asterisk have a "fax" sound tone? If I block their number and play "fax" tone/sound maybe they will remove me from their calling list. I've tried to call them but nothing helps. Any better ideas? They keep calling

I have a subroutine to block spammer by CALLERID(number) exten => 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten => 4,n,Set(goaway=${CALLERID(number):0:2}) exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1) exten => 4,n,GotoIf($["${goaway}" = "V3" ]?blacklisted,s,1) but I just got another spammer (automated calls) who rotates his

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph

Is there any asterisk guru who can explain me how how asterisk knows which context forward the call to? -- Joseph

On Tue, Dec 23, 2014 at 6:34 PM, Rusty Newton <rnewton at digium.com> wrote: > On Tue, Dec 23, 2014 at 4:17 PM, Joseph <syscon780 at gmail.com> wrote: >> Are there any adapters that would allow me to connect asterisk to wifi or we >> are not there yet? >> I have Digium adapter S101i that was discontinued but similar device that >> would connect to wifi

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello,</p><p><br></p><p>Having installed DAHDI to be able to use the meetme() application , when I start the dahdi service it generates me the following error:</p><p>-bash: /etc/init.d/dahdi: No such file or

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello,</p><p>I have not resolved my problem.I renamed my  dahdi file  "mv dahdi.bash dahdi " in the directory /etc/init.d, but it doesn'nt work yet.</p><p>the same error after the command  <strong>/etc/init.d/dahdi

I have followed the instructions in Asterisk The Definitive Guide 4th edition. ?Once I load DAHDI I run the dahdi show channels command and get no such command. ?I have setup all the conf files. ?I compiled DADHI prior to Asterisk. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten =>