similar to: France DID num2sip setup

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

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

On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: > > https://www.voip-info.org/asterisk-security/ > > > > -----Original Message----- > From: asterisk-users [mailto:asterisk-users-bounces at

I wonder if I could have that patch, maybe I could add it to my fail2ban regexp and if you have the correct regexp, I would apperciate that as well. Thanks. On Wed, 29 Aug 2018 19:18:29 -0400, Telium Support Group wrote: > > Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (which has already happened

OK, Thanks. I have a couple of questions -- the line numbers do not match exactly, so can you tell me a couple of lines before and after the line in question? Also, when will this be logged, if its only during sip debug, I need to change it to log when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: > > On 08/29/2018 08:07 PM, John Covici wrote:

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

On 12/30/2017 08:18 PM, Dovid Bender wrote: > 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

Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before

On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group <support at telium.ca> wrote: > Depending on log trolling (Asterisk security log) misses a lot, and also > depends on the SIP/PJSIP folks to not change message structure (which has > already happened numerous time). If you are comfortable hacking > chan_sip.c you may prefer to get the same messages from the AMI. It still

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised 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

Have you enabled DTMF logging and seen the DTMF codes being recognised 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:

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

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On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > Have you enabled DTMF logging and seen the DTMF codes being recognised 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 Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > >

On 12/13/19 9:28 PM, Greg Troxel wrote: > sean darcy <seandarcy2 at gmail.com> writes: > >> I'm moving asterisk to a laptop, so can't use the dahdi board. Is >> there any supported USB dahdi device ? I see the Sangoma USBfxo >> device, but the dahdi driver no longer supports it. Anything else ? > > There is also the ObiHai OBi202 with an OBiLine, which

Call from 'sip' to extension '+1xxxyyyzzzz' rejected because extension not found in context 'out'. But [out] exten => s,1,NoOp( this is the extension: ${EXTEN}) exten => s,n,Answer() exten => s,n(weasels),PlayBack(weasels-eaten-phonesys) ........ If I set "s" to "_." it works. Shouldn't "s" work here? Is it because the

On 12/14/19 11:29 AM, Greg Troxel wrote: > sean darcy <seandarcy2 at gmail.com> writes: > >>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >>> port remoted over SIP. (I am not sure if this is discontinued.) >> >> "FXO port remoted over SIP"? >> >> I have an analog phone system. I can use the obi202 to

Asterisk must have a reasonably large community here in France judging from the number of people who came out to meet Mark. Either that or we were ALL there :) Something I've been waiting for, a voIP carrier on the models we are used to (low monthly or pay as you go, web account) has just set up their first beta test for 1 euro for the first month, 6euros if you decide to keep it. The basic

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ....... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so