search for: seandarcy2

...e-construct itself. I set up a fax-test context: [fax-test] exten=>666,1,NoOp( fax-test ) exten=>666,2,System(/bin/echo "this is a system test"${STRFTIME(${EPOCH},,%H%M)} >> /opt/system-test) exten=>666,3,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email seandarcy2 at gmail -f /var/spool/asterisk/fax/FAXFILE) exten=>666,n,Hangup This works fine on the cli. And /opt/system-test captures the /bin/echo string. AND, the fax2mail log - /var/log/asterisk/faxlog - shows that fax2mail was`called, and there are no errors. So it's not the System() cmd. But...

On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a > particular ip address: > > Retransmitting #10 (NAT) to 5.199.133.128:52734 > <http://5.199.133.128:52734>: >...

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >&g...

...to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear as if the user had not >> dialled any digits. >> >> >> >> >> On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote: >> >>> On 12/20/2014 03:22 PM, sean darcy wrote: >>>> On 12/19/2014 09:42 AM, Rusty Newton wrote: >>>>> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> >>>>> wrote: >>>>>>...

...nised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote: >On 12/20/2014 03:22 PM, sean darcy wrote: >> On 12/19/2014 09:42 AM, Rusty Newton wrote: >>> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> >>>wrote: >>>> I've got a confbridge set up which works if di...

We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ]

...e: > Script kiddies trying to find vulnerable systems that they can make > calls on. Lock down the box with iptables and use fail2ban to block > them. The via is probably bogus unless a box at the DoD was comprimised. > > > > On Sat, Dec 30, 2017 at 6:49 PM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I've been getting a lot of timeouts on non-critical invite > transactions. I turned on sip debug. They were the result of SIP > invites like this: > > Retransmitting #10 (NAT) to 185.107...

...vites to rtp ports ?? > > On 08/29/2018 09:42 AM, Carlos Rojas wrote: >> Hi >> >> Probably somebody is trying to hack your system, you should block that >> ip on your firewall. >> >> Regards >> >> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com >> <mailto:seandarcy2 at gmail.com>> wrote: >> >> I'm getting invites to very high ports every 30 seconds from a >> particular ip address: >> >> Retransmitting #10 (NAT) to 5.199.133.128:52734 >> <http://5.19...

...ax:3] Hangup("DAHDI/4-1", "") in new stack == Spawn extension (incoming-fax, s, 3) exited non-zero on 'DAHDI/4-1' -- Executing [h at incoming-fax:1] System("DAHDI/4-1", "/usr/local/bin/fax2mail.1.sh --dest-name "Sean" --dest-email "seandarcy2 at gmail.com" -f "/var/spool/asterisk/fax/200901141711-0"") in new stack -- Hungup 'DAHDI/4-1' But it doesn't - no email is ever sent. BUT, if I execute the fax2mail cmd from the terminal (pasting from the cli output) it sends the email: /usr/local/bin/fax2m...

On 12/02/2014 02:46 PM, Jeffrey Ollie wrote: > On Tue, Dec 2, 2014 at 1:22 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >> Or do I >> find a new place to put asterisk.pid? > > Also, if you use the native systemd unit file, you no longer need a > PID file, although you still need /run/asterisk to store the control > socket. > So systemd is taking over the g...

On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root. I'm running asterisk under user asterisk. Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asterisk.pid? sean

opusdec -V opusdec opus-tools f2a2e88 (using libopus unknown) I've got an opus file encoded from a .wav off a cd, 44100Hz: opusinfo 2-24-Overture_in_C_\(In_Memoriam\).opus Processing file "2-24-Overture_in_C_(In_Memoriam).opus"... New logical stream (#1, serial: 38134f1f): type opus Encoded with libopus unknown User comments section follows... ENCODER=opusenc from opus-tools

To connect to google voice with xmpp, I've had to turn on the "less secure apps" switch. > You recently changed your security settings so that your Google Account xxxxxxx at gmail.com is no longer protected by modern security standards. > > Please be aware that it is now easier for an attacker to break into your account. My xmpp.conf : type=client

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1&quo...

I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to port to 6111 and now the packets show up. In the server I've set port=6111 in the device in sip.conf, but * is NOT listening for 6111: netstat -an | grep 5060 tcp 0 0

I'm getting invites to very high ports every 30 seconds from a particular ip address: Retransmitting #10 (NAT) to 5.199.133.128:52734: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972 To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748

I'd like to transfer all my pesky telemarketing calls to Jolly Roger . http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html In the middle of a call I'd hit some DTMF sequence, which would dial Jolly Roger and transfer the call after Jolly Roger answers. But blindtransfer requires an extension after you hear "transfer". And I don't

.../2018 09:42 AM, Carlos Rojas wrote: > >> Hi > >> > >> Probably somebody is trying to hack your system, you should block > >> that ip on your firewall. > >> > >> Regards > >> > >> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > >> <mailto:seandarcy2 at gmail.com>> wrote: > >> > >> I'm getting invites to very high ports every 30 seconds from a > >> particular ip address: > >> > >> Retransmitting #10 (NAT) to 5.199.133.128:5273...

...9/2018 09:42 AM, Carlos Rojas wrote: > >> Hi > >> > >> Probably somebody is trying to hack your system, you should block > >> that ip on your firewall. > >> > >> Regards > >> > >> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > >> <mailto:seandarcy2 at gmail.com>> wrote: > >> > >> I'm getting invites to very high ports every 30 seconds from a > >> particular ip address: > >> > >> Retransmitting #10 (NAT) to 5.199.133.128:52734...

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A