Re: [asterisk-users] Block Spam Calls

D'Arcy Cain writes: > Not bad. I was toying with another idea. I find that if I don't answer > a robot fast enough it just hangs up. How about ring two or three times > before passing to the actual extension? You could try that and let us know, but I suspect: some robocallers don't hang

Re: [asterisk-users] Block Spam Calls

On 12/12/19 5:33 PM, Greg Woods wrote: > Most spam calls are robocalls these days. At my house, I can block > pretty much all of the robocalls by requiring the caller to take some > action before ringing the phones. In our case, the action is just to > dial 1 for my wife or 2 for me. The only

Re: [asterisk-users] Block Spam Calls

Most spam calls are robocalls these days. At my house, I can block pretty much all of the robocalls by requiring the caller to take some action before ringing the phones. In our case, the action is just to dial 1 for my wife or 2 for me. The only difference it makes in the end is which voice

Re: [asterisk-users] Block Spam Calls

This is exactly what I do - “press 1 for a human” Works great From: asterisk-users on behalf of Greg Woods Sent: Thursday, December 12, 2019 6:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block Spam Calls Most

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=-

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 7:57 AM marek wrote: > with wireshark i need decrypt traffic every call which is time consuming. > get debug from pjnat through asterisk is not possible because of technical > reasons or nobody did it? > > > in my case its strange that ice candidates are the same > > good

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 8:57 AM marek wrote: > Asterisk is on public IP (as described in the first email) > > i have 10 years experience in voip, 4 years webrtc in production. i know > about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism > > but i confess. i dont understand

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP.

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 6:39 AM marek wrote: > hi, > > i have following topology > > PSTN - Asterisk internet - router - jssip client (wss) > > Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP > connection to PSTN > > router - public IP/private IP (NAT) > > jssip

[asterisk-users] asterisk pjsip webrtc rtp to private IP

hi, i have following topology PSTN - Asterisk internet -  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :  

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 10:57 AM marek wrote: > thank you very much. this is exactly whats needed for debug > > example output for your info > [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : > icess0x7f5d44081e88 .Added new remote candidate from the request: > 2.2.2.2:57536 > [Dec 12

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 11:40 AM marek wrote: > examples of "interesting" information like ICE result and howto make > "minimal" configuration of pjproject.conf > > i.e. > > for debugging app_queue.so > > core set debug 5 app_queue.so > > for debugging RTP > > core set debug 10 rtp_engine >

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

https://issues.asterisk.org/jira/browse/ASTERISK-28656 Dne 12/12/2019 v 16:49 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 11:40 AM marek > wrote: examples of "interesting" information like ICE result and howto make "minimal" configuration of

Re: [asterisk-users] Block Spam Calls

Title: Re: [asterisk-users] Block Spam Calls Hello Alexander, Tuesday, December 10, 2019, 7:57:54 AM, you wrote: Hi All.  Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller?  I ask because

Re: [asterisk-users] Site to site VPN problems

Title: Re: [asterisk-users] Site to site VPN problems Hello Jan, Tuesday, December 3, 2019, 8:49:28 PM, you wrote: Jan> The next thing to look at is firewall rules. So it wasn't the firewall. I eventually fixed it by creating 512 entries for Twilio so no matter what IP they sent it from it

Re: [asterisk-users] Site to site VPN problems

Ira, What version of Asterisk are you using, and what channel driver? There has to be a better way than to create hundreds of peer entries. On Thu, Dec 12, 2019 at 12:26 PM Ira wrote: > Hello Jan, > > Tuesday, December 3, 2019, 8:49:28 PM, you wrote: > > Jan> The next thing to look at is

Re: [asterisk-users] Site to site VPN problems

You only need to add the Signaling IPs as peers. The Media IPs only need to be whitelisted in your firewall. Furthermore, you can actually trim the list down to only one region by following their instructions here https://www.twilio.com/docs/sip-trunking#OriginationURI-region On Thu, 12 Dec 2019