Hi Lonnie Sorry for the miscommunication. I have tried both already. Maybe I have a faulty Ethernet port or driver?
Regards Michael Knill -----Original Message----- From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> Date: Tuesday, 26 December 2017 at 1:00 pm To: AstLinux List <astlinux-users@lists.sourceforge.net> Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk Hi Michael, Give #2 a try again. If still no go, try this: -- arno-iptables-firewall stop ; arno-iptables-firewall start -- Lonnie On Dec 25, 2017, at 7:53 PM, Michael Knill <michael.kn...@ipcsolutions.com.au> wrote: > Ok its happened again and I have an opportunity to do some troubleshooting. > Unfortunately I have not been able to get it reachable after trying the > following: > 1) Add a discrete firewall rule for 5060 from the IP Address Pass Ext -> Local > 2) Stop Asterisk, wait for 3 minutes and start Asterisk > 3) Reload config > > Sip debug is multiple options pings with no response: > OPTIONS sip:103.226.10.78 SIP/2.0. > Via: SIP/2.0/UDP 124.148.24.56:5060;branch=z9hG4bK56ffba00. > Max-Forwards: 70. > From: "asterisk" <sip:asterisk@124.148.24.56>;tag=as1daab990. > To: <sip:103.226.10.78>. > Contact: <sip:asterisk@124.148.24.56:5060>. > Call-ID: 21e8014648d8d5d92f4051c42ff6dda3@124.148.24.56:5060. > CSeq: 102 OPTIONS. > User-Agent: IBCCM. > Date: Tue, 26 Dec 2017 01:50:49 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE. > Supported: replaces. > Content-Length: 0. > > U 2017/12/26 12:50:52.309751 124.148.24.56:5060 -> 103.226.10.78:5060 > > OPTIONS sip:103.226.10.78 SIP/2.0. > Via: SIP/2.0/UDP 124.148.24.56:5060;branch=z9hG4bK56ffba00. > Max-Forwards: 70. > From: "asterisk" <sip:asterisk@124.148.24.56>;tag=as1daab990. > To: <sip:103.226.10.78>. > Contact: <sip:asterisk@124.148.24.56:5060>. > Call-ID: 21e8014648d8d5d92f4051c42ff6dda3@124.148.24.56:5060. > CSeq: 102 OPTIONS. > User-Agent: IBCCM. > Date: Tue, 26 Dec 2017 01:50:49 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE. > Supported: replaces. > Content-Length: 0. > > U 2017/12/26 12:50:53.309434 124.148.24.56:5060 -> 103.226.10.78:5060 > > OPTIONS sip:103.226.10.78 SIP/2.0. > Via: SIP/2.0/UDP 124.148.24.56:5060;branch=z9hG4bK56ffba00. > Max-Forwards: 70. > From: "asterisk" <sip:asterisk@124.148.24.56>;tag=as1daab990. > To: <sip:103.226.10.78>. > Contact: <sip:asterisk@124.148.24.56:5060>. > Call-ID: 21e8014648d8d5d92f4051c42ff6dda3@124.148.24.56:5060. > CSeq: 102 OPTIONS. > User-Agent: IBCCM. > Date: Tue, 26 Dec 2017 01:50:49 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE. > Supported: replaces. > Content-Length: 0. > > U 2017/12/26 12:51:03.309654 124.148.24.56:5060 -> 103.226.10.78:5060 > > OPTIONS sip:103.226.10.78 SIP/2.0. > Via: SIP/2.0/UDP 124.148.24.56:5060;branch=z9hG4bK40f34f93. > Max-Forwards: 70. > From: "asterisk" <sip:asterisk@124.148.24.56>;tag=as47f11228. > To: <sip:103.226.10.78>. > Contact: <sip:asterisk@124.148.24.56:5060>. > Call-ID: 670c60845c515149354254b721550d6b@124.148.24.56:5060. > CSeq: 102 OPTIONS. > User-Agent: IBCCM. > Date: Tue, 26 Dec 2017 01:51:03 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE. > Supported: replaces. > Content-Length: 0. > > Its still broken so any further ideas for testing before I reboot? > > Regards > Michael Knill > > -----Original Message----- > From: Michael Knill <michael.kn...@ipcsolutions.com.au> > Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> > Date: Tuesday, 19 December 2017 at 12:42 pm > To: AstLinux List <astlinux-users@lists.sourceforge.net> > Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk > > Cool thanks Lonnie > > Looks like I will be adding some firewall rules to my default build. > > Regards > Michael Knill > > -----Original Message----- > From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> > Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> > Date: Tuesday, 19 December 2017 at 12:50 pm > To: AstLinux List <astlinux-users@lists.sourceforge.net> > Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk > > Michael, > > You stated the qualify yes/no tradeoffs perfectly. > > Personally I would use "no" if it is not needed, but if you like the latency > check give "yes" a shot - the chance of messing-up a firewall state along the > way should be low. > > Lonnie > > > On Dec 18, 2017, at 7:37 PM, Michael Knill > <michael.kn...@ipcsolutions.com.au> wrote: > >> Thanks Lonnie >> >> Is there a reason why I would turn off qualify? Its kind of nice knowing >> whether your ITSP peer is up and it also can highlight potential network >> issues when you start getting lags and unreachables! >> >> But if there is a risk of it messing with the Firewall state then it may be >> worth doing. >> >> Regards >> Michael Knill >> >> -----Original Message----- >> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> >> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> >> Date: Tuesday, 19 December 2017 at 11:50 am >> To: AstLinux List <astlinux-users@lists.sourceforge.net> >> Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk >> >> Hi Michael, >> >> My commercial SIP provider offers exactly the scenario I suggested, using >> static IPv4 authentication as you are doing. >> >> I used the same suggested setup for a dynamic remote trunk for years ... >> until I switched it to WireGuard a few weeks ago. Though it was a dynamic >> cable modem, not PPPoE. >> >> But, if you are having the same PPPoE issue with a commercial SIP provider, >> for your clients trying "qualify=no" and opening UDP 5060 from your IPv4 >> server host would be worth a try. >> >> Though going forward, if the server's IP changes you will have more editing >> to do for each remote client. >> >> Lonnie >> >> >> On Dec 18, 2017, at 6:11 PM, Michael Knill >> <michael.kn...@ipcsolutions.com.au> wrote: >> >>> Hi Lonnie >>> >>> In this case it is an Astlinux box at the other end but I also have issues >>> with SIP Trunk providers that I have no administrative control over. >>> Interestingly I don't think this happens when you lose the PPPoE through >>> network or Layer 1 issues. For this one it was LCP terminated by Peer which >>> may do something differently. >>> >>> I was thinking therefore of adding a 5060 firewall rule for all my clients >>> using the SIP Provider address ranges. Do you think that may solve this >>> issue e.g. the firewall rule is static and not dynamic? >>> Virtually all my systems have a static IP. >>> >>> Regards >>> Michael Knill >>> >>> -----Original Message----- >>> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> >>> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> >>> Date: Tuesday, 19 December 2017 at 9:22 am >>> To: AstLinux List <astlinux-users@lists.sourceforge.net> >>> Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk >>> >>> Hi Michael, >>> >>> What you are currently doing works *almost* all the time, but there seems >>> to be edge conditions when the PPPoE link cycles. >>> >>> If you don't need the server end "qualify=yes" to monitor the latency >>> and/or immediate "channel not available" (which can also cause issues) ... >>> I would consider "qualify=no" at the server end and add a UDP 5060 firewall >>> rule for each remote static IPv4 SIP client. This also documents the SIP >>> endpoints, easily disable one at the network level, etc. . >>> >>> Additionally, if the remote SIP clients are dynamic (w/ Dynamic DNS Update) >>> and register with asterisk to authenticate, at the server end you can use >>> the AIF dyndns-host-open plugin and define >>> -- >>> DYNDNS_HOST_OPEN_UDP="remote1.sip.tld~5060 remote2.sip.tld~5060" >>> -- >>> for every remote dynamic client. >>> >>> In all cases the remote SIP clients (static or dynamic) are configured with >>> "qualify=yes" at the remote end to open their firewall. >>> >>> Possibly you can test with one trunk and see if it helps. >>> >>> Lonnie >>> >>> >>> >>> On Dec 18, 2017, at 3:23 PM, Michael Knill >>> <michael.kn...@ipcsolutions.com.au> wrote: >>> >>>> Awesome sipgrep! I keep finding new tools all the time that I wished I >>>> knew about earlier. Maybe we should have a helpful Astlinux tools page one >>>> day. >>>> >>>> Both ends use qualify (SIP OPTIONS) so the firewall state should always >>>> remain open but do you think that adding a static rule in the firewall >>>> more reliable? >>>> I was thinking of doing this for all my systems for security purposes but >>>> it may also remove the reliance on the firewall setting up dynamic states! >>>> >>>> Regards >>>> Michael Knill >>>> >>>> -----Original Message----- >>>> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> >>>> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> >>>> Date: Tuesday, 19 December 2017 at 1:54 am >>>> To: AstLinux List <astlinux-users@lists.sourceforge.net> >>>> Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk >>>> >>>> Hi Michael, >>>> >>>>> The terminating SIP server is actually another Astlinux box. >>>> >>>> Cool, so this should be solvable. >>>> >>>> Yes, as you suggested a sip debug on the Asterisk CLI (or use "sipgrep") >>>> would help ... if you can VPN into the server also a help during the >>>> failure. >>>> >>>> I would try setting qualify=yes to only the client end SIP trunk, not the >>>> server end, this would consistently establish new firewall states from the >>>> client to the server endpoint. I'm assuming the client end does not open >>>> UDP 5060 and relies on the outbound SIP OPTIONS traffic to establish the >>>> firewall state. >>>> >>>> Additionally, since your SIP trunk is authenticating on the static IPv4 >>>> address of the PPPoE endpoint, make sure that IPv4 address is indeed >>>> static during any PPPoE hiccups. >>>> >>>> Lonnie >>>> >>>> PS, this is a perfect scenario for placing the SIP trunk over a WireGuard >>>> VPN, for your future setup down the road :-) >>>> >>>> >>>> >>>> On Dec 17, 2017, at 10:56 PM, Michael Knill >>>> <michael.kn...@ipcsolutions.com.au> wrote: >>>> >>>>> Hi Lonnie >>>>> >>>>> No it's an IP Address only SIP Trunk so no registration. Only Options >>>>> Pings. I should have done a sip debug on the Asterisk CLI I think. >>>>> It's a static local IP Address (passed by PPPoE) and it did not change. >>>>> The terminating SIP Server has a single Public IP Address so there should >>>>> be no NAT but I cannot guarantee. The terminating SIP server is actually >>>>> another Astlinux box. >>>>> >>>>> Im just wondering; if no IP Address or Port had changed, shouldn't the >>>>> firewall state remain the same anyway? >>>>> >>>>> I will try that. More testing required! >>>>> >>>>> Regards >>>>> Michael Knill >>>>> >>>>> -----Original Message----- >>>>> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com> >>>>> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net> >>>>> Date: Monday, 18 December 2017 at 3:39 pm >>>>> To: AstLinux List <astlinux-users@lists.sourceforge.net> >>>>> Subject: Re: [Astlinux-users] Problems with PPPoE and Asterisk >>>>> >>>>> Hi Michael, >>>>> >>>>> I see your logs stopped after the PPPoE connection was reestablished. >>>>> Are there a bunch of asterisk register timeouts after that point ? >>>>> >>>>> Does your PPPoE "local IP address" typically change every time the >>>>> connection goes down and back up ? >>>>> >>>>> This does sound like a firewall invalid state issue and rebooting allows >>>>> the invalid state's TTL to expire, BUT the firewall state is probably not >>>>> in AstLinux but rather upstream. >>>>> >>>>> Is it possible there is NAT in the path between your PPPoE "local IP >>>>> address" and the SIP server ? >>>>> >>>>> Does the remote SIP server resolve to a single IPv4 address, or is it a >>>>> round-robin of IPv4 addresses ? >>>>> >>>>> >>>>>> Yes I agree. I think it's a firewall problem although a Restart Firewall >>>>>> did not fix it ( >>>>>> Can you think of any good debugging I can do if it happens again? >>>>> >>>>> The next time, rather than rebooting, I would try from the CLI ... >>>>> -- >>>>> service asterisk stop >>>>> (wait 3 minutes or more - use your watch) >>>>> service asterisk init >>>>> -- >>>>> If that re-establishes SIP connectivity, that would imply the >>>>> stuck-firewall-state was upstream. >>>>> >>>>> Lonnie >>>>> >>>>> >>>>> >>>>> On Dec 17, 2017, at 4:43 PM, Michael Knill >>>>> <michael.kn...@ipcsolutions.com.au> wrote: >>>>> >>>>>> Hi Group >>>>>> >>>>>> I am still having issues with PPPoE and Asterisk connectivity. >>>>>> >>>>>> This happened over the weekend with one of my sites: >>>>>> Dec 17 00:30:09 2005-Shaw_BG-CM1 local0.notice asterisk[1266]: >>>>>> NOTICE[1408]: chan_sip.c:30077 in sip_poke_noanswer: Peer 'cts_trunk' is >>>>>> now UNREACHABLE! Last qualify: 33 >>>>>> Dec 17 00:30:33 2005-Shaw_BG-CM1 local0.notice asterisk[1266]: >>>>>> NOTICE[1408]: chan_sip.c:24558 in handle_response_peerpoke: Peer >>>>>> 'cts_trunk' is now Reachable. (34ms / 2000ms) >>>>>> Dec 17 00:30:52 2005-Shaw_BG-CM1 daemon.info pppd[336]: LCP terminated >>>>>> by peer >>>>>> Dec 17 00:30:52 2005-Shaw_BG-CM1 daemon.info pppd[336]: Connect time >>>>>> 568.6 minutes. >>>>>> Dec 17 00:30:52 2005-Shaw_BG-CM1 daemon.info pppd[336]: Sent 1190012 >>>>>> bytes, received 1282467 bytes. >>>>>> Dec 17 00:30:55 2005-Shaw_BG-CM1 daemon.notice pppd[336]: Connection >>>>>> terminated. >>>>>> Dec 17 00:30:55 2005-Shaw_BG-CM1 daemon.notice pppd[336]: Modem hangup >>> >>> >>> ------------------------------------------------------------------------------ >>> Check out the vibrant tech community on one of the world's most >>> engaging tech sites, Slashdot.org! http://sdm.link/slashdot >>> _______________________________________________ >>> Astlinux-users mailing list >>> Astlinux-users@lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/astlinux-users >>> >>> Donations to support AstLinux are graciously accepted via PayPal to >>> pay...@krisk.org. >>> >>> >>> ------------------------------------------------------------------------------ >>> Check out the vibrant tech community on one of the world's most >>> engaging tech sites, Slashdot.org! http://sdm.link/slashdot >>> _______________________________________________ >>> Astlinux-users mailing list >>> Astlinux-users@lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/astlinux-users >>> >>> Donations to support AstLinux are graciously accepted via PayPal to >>> pay...@krisk.org. >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Check out the vibrant tech community on one of the world's most >> engaging tech sites, Slashdot.org! http://sdm.link/slashdot >> _______________________________________________ >> Astlinux-users mailing list >> Astlinux-users@lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/astlinux-users >> >> Donations to support AstLinux are graciously accepted via PayPal to >> pay...@krisk.org. >> >> >> ------------------------------------------------------------------------------ >> Check out the vibrant tech community on one of the world's most >> engaging tech sites, Slashdot.org! http://sdm.link/slashdot >> _______________________________________________ >> Astlinux-users mailing list >> Astlinux-users@lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/astlinux-users >> >> Donations to support AstLinux are graciously accepted via PayPal to >> pay...@krisk.org. >> >> > > > ------------------------------------------------------------------------------ > Check out the vibrant tech community on one of the world's most > engaging tech sites, Slashdot.org! http://sdm.link/slashdot > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org. > > > ------------------------------------------------------------------------------ > Check out the vibrant tech community on one of the world's most > engaging tech sites, Slashdot.org! http://sdm.link/slashdot > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org. > > > ------------------------------------------------------------------------------ > Check out the vibrant tech community on one of the world's most > engaging tech sites, Slashdot.org! http://sdm.link/slashdot > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org. ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org. ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
