[asterisk-users] issue with NAT
First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: [TK]D-Fender Note for elfranne's situation, : nat=force_rport,comedia should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with NAT
Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens: First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: [TK]D-Fender Note for elfranne's situation, : nat=force_rport,comedia should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu Hi Tom, you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents. read more about STUN at: http://www.voip-info.org/wiki/view/STUN and there is a list of public STUN Server. Regards -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with NAT
Am 03.11.2014 um 13:47 schrieb Rainer Piper: Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens: First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: [TK]D-Fender Note for elfranne's situation, : nat=force_rport,comedia should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu Hi Tom, you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents. read more about STUN at: http://www.voip-info.org/wiki/view/STUN and there is a list of public STUN Server. Regards the add path header support in chan_sip could help as well. look at https://issues.asterisk.org/jira/browse/ASTERISK-16884 [Test danes 202] ... ... nat=force_rport,comedia usepath=yes ... ... [test danes 203] ... ... nat=force_rport,comedia usepath=yes ... ... -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sslv3 alert handshake failure error
Hi Jeff, Thanks for the response. I am using PJSIP soft clients and PJSIP uses TLSv1 by default. Even on network traces (using wireshark), I can observed TLSv1 protocol is being used. I am not getting why it is falling back on sslv3. Could you please guide me here? Polease correct me in case I miss something here. More-ever, I have something as following in extensions.conf exten = 100,1,Answer() same = n,Wait(1) same = n,Playback(hello-world) same = n,Hangup() And call to 100 (Req URI - INVITE sips:1...@pbx.asterisk1.org;trasnport=tls SIP/2.0) from either of PJSIP soft clients works perfectly. So I wonder, how it works here and it fails when I dial an extension configured on a soft phones -- Thanks Atul Thosar On 2 November 2014 22:50, Jeffrey Walton noloa...@gmail.com wrote: == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Nov 2 21:20:05] WARNING[3571]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! It sounds like SSLv3 is being used by one of the endpoints. SSLv3 is broken. Its been known broken for about 10 years. Its been more broken recently (???). It should not have been used previous to POODLE, and it should not be used now. And don't use that crap UA's came up with (TLS_FALLBACK_SCSV). Always advertise the protocols you are willing to accept, and don't fallback to insecure protocols. My protocol selections are TLS 1.0, 1.1 and 1.2. I allow TLS 1.0 for interoperability, but I'd like to bury it too. If you control the server and the clients, then you should be able to safely kill-off TLS 1.0 since interop is not a concern. Jeff On Sun, Nov 2, 2014 at 11:35 AM, Atul Thosar atultho...@gmail.com wrote: Hi All, I am using asterisk-11.12.0 version and I am trying to setup secure call (TLS + SRTP) between two extensions and while making a call, I got following error *CLI == Using SIP RTP CoS mark 5 -- Executing [6004@from-office:1] Dial(SIP/6003-, SIP/6004,20) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/6004 SSL certificate ok == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Nov 2 21:20:05] WARNING[3571]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! I followed instruction given in https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial;, but no luck. I googled around the issue and found solution mentioned by Patrick ( https://www.mail-archive.com/asterisk-users@lists.digium.com/msg274038.html ) Did anyone has tried this solution and found it is working? I tried to create certificates with keyUsage/extendedKeyUsage, but it is not working. I have one more query - When the SIP user agents are able to register successfully with TLS, why more handshake is required while making a call? Can't Asterisk use existing TLS connection with Leg B to forward INVITE request? Could anyone please educate me on the same? I am little confused here. Thanks in advance. -- Atul Thosar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with NAT
On Mon, Nov 3, 2014 at 6:58 AM, Rainer Piper rainer.pi...@soho-piper.de wrote: Am 03.11.2014 um 13:47 schrieb Rainer Piper: Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens: First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: [TK]D-Fender Note for elfranne's situation, : nat=force_rport,comedia should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu Hi Tom, you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents. read more about STUN at: http://www.voip-info.org/wiki/view/STUN and there is a list of public STUN Server. Regards the add path header support in chan_sip could help as well. look at https://issues.asterisk.org/jira/browse/ASTERISK-16884 [Test danes 202] ... ... nat=force_rport,comedia usepath=yes ... ... [test danes 203] ... ... nat=force_rport,comedia usepath=yes ... ... Path support will only help if there are intermediary proxies, and even then won't help with media (assuming OP meant 'no sound'). I could have missed it in the pastebin, but I didn't see a request/response from Asterisk that was either sent to a private IP address or contained a private IP address in the SDP. In the trace that you provided, which request/response did you feel was in error? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 12 - zombie processes
Hello Asterisk users developers, I have opened an issue few days ago regarding the crash and the zombie processes. I haven't received any response and it has't been assigned. If something is wrong or missing with the issue please get back to me and I will handle it. Please look into it because we still have crashes every day or two, and we can't reproduce the issue in our lab with a simulator. Thank you, Yaron. On Thu, Oct 30, 2014 at 9:01 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello everyone, I have opened a ticket number - ASTERISK-24471 https://issues.asterisk.org/jira/browse/ASTERISK-24471. I have attached the backtrace of the core file. The backtrace was taken on the server running 12.6.1. If you need any information please get back to me. Thank you. Yaron. On Thu, Oct 30, 2014 at 8:43 AM, Yaron Nachum nachum.ya...@gmail.com wrote: No, I went over all my scripts. Thanks for the help. Yaron On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Mathew, When I run 'ps -ef|grep asterisk' the following processes are displayed: root 6861 1 0 Aug27 ?00:00:00 /bin/sh /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf asterisk 8062 6861 3 Oct27 ?00:44:56 /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c root 20776 2200 0 11:20 pts/200:00:33 tail -f asterisk.log asterisk 23076 8062 0 17:01 ?00:00:00 [asterisk] defunct asterisk 23897 8062 0 17:03 ?00:00:00 [asterisk] defunct also when I run top the same amount of zombie processes are displayed: Tasks: 185 total, 1 running, 182 sleeping, 0 stopped, 2 zombie Regarding the AGI - we are using AGI in order to run php scripts for external logic. I have printed the PIDs of the php scripts and none of them are related to the PID's of those zombie processes. Do you have any idea how to find out what are these processes? Yaron. Are you doing anything like: # asterisk -rx 'core show channels' via an external process? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
