| 00:03.04 | *** part/#asterisk kharwell (kharwell@nat/digium/x-vnamnvsbxromlzyb) |
| 00:23.39 | *** join/#asterisk craigify (~craigify@162.216.46.155) |
| 00:35.33 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 01:01.13 | *** join/#asterisk KaliLinuxGR (~KaliLinux@unaffiliated/kalilinuxgr) |
| 02:11.19 | *** join/#asterisk eaxxae (~eaxxae@unaffiliated/eaxxae) |
| 02:24.57 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d20:239:11e0:707c:2961:d41e) |
| 03:10.15 | *** join/#asterisk eaxxae (~eaxxae@unaffiliated/eaxxae) |
| 03:35.47 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 03:45.18 | *** join/#asterisk _AxS_ (~axs@gentoo/developer/axs) |
| 03:58.35 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 03:59.13 | *** join/#asterisk _AxS_ (~axs@gentoo/developer/axs) |
| 04:17.48 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-iickwpqkbswbwfsg) |
| 04:20.13 | *** join/#asterisk _AxS_ (~axs@gentoo/developer/axs) |
| 04:21.55 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 04:33.02 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 04:52.06 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 05:11.53 | *** join/#asterisk kschreyack (~klaus@vpn.myexpertise.net) |
| 05:13.33 | kschreyack | Good evening everyone. Anyone ever seen asterisk successfully register its trunks and then later start timing out? If I restore to a previous snap all is well, but then it occurs again after time. Strange problem for me.. |
| 06:03.23 | *** join/#asterisk J0hnSteel (~J0hnSteel@92.55.116.125) |
| 06:21.42 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 06:39.30 | *** join/#asterisk spicyramen_ (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 07:04.39 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 07:23.12 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 07:32.58 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 07:40.27 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 07:42.51 | *** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta) |
| 08:25.09 | *** join/#asterisk eaxxae (~eaxxae@unaffiliated/eaxxae) |
| 08:32.15 | *** join/#asterisk Jack17 (~jackkk@212.34.23.49) |
| 08:33.16 | Jack17 | Anyway to do a timeout function , im using asterisk 13 I want to do function withtimeout response time is 3 sec if no reply goto another context |
| 08:33.20 | Jack17 | possible ? |
| 08:34.18 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 08:43.59 | [TK]D-Fender | of course |
| 08:54.27 | Jack17 | Would you provide me with a command tk defender |
| 08:54.31 | Jack17 | [TK]D-Fender |
| 08:54.50 | [TK]D-Fender | "core show function TIMEOUT" |
| 08:56.27 | Jack17 | alright what would be the return value |
| 08:56.50 | [TK]D-Fender | There is no return |
| 08:56.57 | [TK]D-Fender | read its instructions |
| 08:57.36 | Jack17 | ok how to jump to another area incase of no response |
| 08:57.42 | Jack17 | context i mean |
| 08:58.36 | [TK]D-Fender | set your timout. in the timeout exten do your goto |
| 09:00.50 | Jack17 | exten => s,n,Set(TIMEOUT(response)=3) |
| 09:01.29 | Jack17 | exten => t,n,Goto(s,1) |
| 09:01.31 | Jack17 | like that |
| 09:02.12 | [TK]D-Fender | except for the clear priority mistake, yes |
| 09:02.53 | [TK]D-Fender | and that you're no leaving the context like you said you wanted to do |
| 09:02.58 | [TK]D-Fender | not |
| 09:05.52 | Jack17 | the exten => t,n,Goto(s,1) will be executed if no response |
| 09:06.10 | [TK]D-Fender | [TK]D-Fender> except for the clear priority mistake, yes |
| 09:06.23 | [TK]D-Fender | as I see now it will not get executed ever |
| 09:07.01 | Jack17 | same => n,Goto(br) |
| 09:07.10 | [TK]D-Fender | no |
| 09:07.18 | [TK]D-Fender | getting worse |
| 09:07.34 | Jack17 | same => t,n,Goto(br) |
| 09:07.46 | [TK]D-Fender | even worse |
| 09:16.47 | *** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com) |
| 09:26.47 | Jack17 | ~pb |
| 09:26.48 | infobot | pastebin is probably a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 09:36.04 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:2d15:4cf0:73e6:ff39) |
| 09:46.20 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:d20:90a7:57a4:c50) |
| 10:19.18 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:40.53 | *** join/#asterisk chendy (~chendy@120.237.116.82) |
| 10:43.36 | *** join/#asterisk chendy_ (~chendy@117.136.79.153) |
| 10:58.44 | *** join/#asterisk iheartlinux (~jwpierce3@mail.johnthecomputerguy.com) |
| 11:47.58 | *** join/#asterisk slack (~slack@88-111-150-65.dynamic.dsl.as9105.com) |
| 12:00.44 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 12:23.02 | *** join/#asterisk avb (~avb@45.55.203.176) |
| 12:56.45 | *** join/#asterisk MajesticFudgie (~MajesticF@host81-135-241-192.range81-135.btcentralplus.com) |
| 12:56.55 | MajesticFudgie | Can someone remind me the correct command to turn debugging on |
| 12:59.07 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 13:01.27 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:894d:9f60:bb2b:d0a5) |
| 13:09.41 | Jack17 | rasterisk -vvvvvvvvvvr |
| 13:13.36 | MajesticFudgie | Doesn;t tell me why calls arent going through though |
| 13:14.22 | MajesticFudgie | I thought the sip debug was "sip set debug on" but "sip" doesnt exist |
| 13:38.38 | MajesticFudgie | Disregard that. It seems updating my linux distro caused asterisk to insist that its modules were build for an older version. Updating my asterisk and recompiling fixed it :P |
| 13:49.01 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:85a8:8de6:b419:e69d) |
| 14:01.18 | MajesticFudgie | Is it possible to set the onhold music for a specific extension? Rather than the same onhold music for all? |
| 14:29.29 | Samot | Are you using Chan_SIP or Chan_PJSIP for your devices/peers? |
| 14:32.30 | Samot | And yes, you can specify MoH per peer |
| 14:37.53 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-pqvdexygdjtkydkp) |
| 14:49.48 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:819f:db3:23e7:a719) |
| 14:51.16 | WIMPy | You can even set it per extension. |
| 14:56.06 | Samot | That's what I meant as well. |
| 14:56.12 | Samot | device/peer.. |
| 14:57.56 | WIMPy | You might be able to do that as well. |
| 15:00.08 | *** join/#asterisk shymega (~shymega@shymega.vps.bitfolk.com) |
| 15:25.58 | *** join/#asterisk KaliLinuxGR (~KaliLinux@unaffiliated/kalilinuxgr) |
| 15:41.37 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-djgavhqvjeywyljw) |
| 15:50.55 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:e469:e63:ef67:9958) |
| 16:17.37 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 16:36.32 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 16:51.35 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:c8a3:96b0:c966:aaea) |
| 16:53.57 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 17:04.07 | *** join/#asterisk drjfreeze (~jimfreeze@107-216-41-190.lightspeed.austtx.sbcglobal.net) |
| 17:04.46 | drjfreeze | Hi. We just converted over to TWC SIP for our voice provider. |
| 17:05.26 | drjfreeze | All looks like it works well, except that calls coming in (2nd caller or more), there is no audio |
| 17:05.57 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 17:06.20 | drjfreeze | Anyone have an idea of why subsequent calls would not be receiving audio on a SIP configuration? |
| 17:15.46 | [TK]D-Fender | look at the call] |
| 17:25.58 | drjfreeze | [TK]D-Fender: TWC has an SBC which is what I think is the issue |
| 17:26.20 | drjfreeze | the SBC is local to asterisk box |
| 17:36.14 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 17:36.17 | *** join/#asterisk ghenry (~ghenry@openldap/ghenry) |
| 17:38.24 | [TK]D-Fender | look at the call |
| 17:41.48 | drjfreeze | Not sure if this matters, but I see the following: <sip:15123517454@192.168.121.50> |
| 17:42.31 | [TK]D-Fender | You need to look at the ENTIRE thing |
| 17:42.38 | drjfreeze | the 121.50 is the IP of the asterisk box that faces the internal phones, but the IP that faces the sip is 192.168.0.10 |
| 17:42.44 | [TK]D-Fender | not just one individual line |
| 17:43.16 | [TK]D-Fender | Thazt isn't even showing which header we're loking at, where anything is sourcing from, etc |
| 17:43.30 | drjfreeze | what commands should I use to get the info you are looking for? |
| 17:48.44 | [TK]D-Fender | Depends which channel driver you are using |
| 17:48.58 | [TK]D-Fender | chan_sip = sip set debug on |
| 17:49.12 | [TK]D-Fender | chan_pjsip = pjsip set logger on |
| 17:52.15 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:ec5f:3fb4:38e1:8b0b) |
| 18:46.40 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 18:49.21 | dan_j | Hi. Does anyone know if the Panasonic KX-UT models support TLS and has anyone managed to connect them up to Asterisk while using TLS? |
| 18:54.35 | [TK]D-Fender | what does their manual & data sheet say? |
| 18:56.49 | dan_j | Not a word. Just can't imagine that a fairly new model phone doesn't support TLS |
| 18:57.18 | [TK]D-Fender | if it's ot in the manual, it is what it is |
| 18:58.29 | dan_j | hmm |
| 19:04.27 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:05.02 | dan_j | [TK]D-Fender: Can asterisk operate with both TLS and standard sip at the same time? Or is it either/or? |
| 19:05.15 | Samot | Both. |
| 19:05.20 | Samot | As well as WS/WSS |
| 19:05.40 | Samot | A device can use any or all transports you want it to. |
| 19:05.57 | dan_j | ? |
| 19:06.12 | Samot | You can do transport=udp,tcp,tls |
| 19:06.18 | Samot | And it will work on all three. |
| 19:06.30 | Samot | If setup correctly. |
| 19:06.35 | dan_j | Ah. But the endpoint still has to support TLS in order to us it? Or is it part of the SIP standard? |
| 19:06.50 | WIMPy | LOL |
| 19:06.53 | Samot | Yes, the device has to support TLS in order to do TLS. |
| 19:07.02 | Samot | Much like it would need to support WS/WSS to do those. |
| 19:07.26 | [TK]D-Fender | Every peer is whatever the hell you tell it to be |
| 19:07.35 | dan_j | Annoyingly, the specs say that the phone complies with the SIP RFC which includes TLS support, but no mention of whether it actually supports TLS |
| 19:07.54 | [TK]D-Fender | Was the "transport" parameter in the peer shown in the sample config not clear enough? |
| 19:08.11 | Samot | Just because it's RFC compliant doesn't mean it supports all the RFC features. |
| 19:08.11 | WIMPy | If it doesn't say so, it probably doesn't. |
| 19:08.29 | dan_j | WIMPy: nuts |
| 19:08.39 | dan_j | ok, thanks for the help. |
| 19:08.39 | [TK]D-Fender | Why are we asking this question backwards? |
| 19:08.49 | [TK]D-Fender | Do you SEE all the setting on the phone? |
| 19:09.05 | [TK]D-Fender | Does the manual show you all the parameters to set? |
| 19:09.09 | [TK]D-Fender | THAT is what you have |
| 19:09.42 | dan_j | Yes. No mention in the example config files. But I have been given some additional config by my distributor which is not in the manual nor the example config files. Hence why i'm asking if anyones done it. |
| 19:09.51 | [TK]D-Fender | If it doesn't offer you a choice of transport as a field then it only supports ONE |
| 19:10.01 | dan_j | It's the weekend, so I'm not going to get a response from them till monday. |
| 19:10.03 | [TK]D-Fender | Which I doubt would be TLS in place of UDP |
| 19:10.09 | [TK]D-Fender | From who? |
| 19:10.19 | [TK]D-Fender | You don't have the manuals DOWNLAODED off their website? |
| 19:10.41 | [TK]D-Fender | Tell me you're not relying on a CLIENT to interpret this from miles away.... |
| 19:11.14 | dan_j | [TK]D-Fender: They manual doesnt include some settings, as demonstrated by some additional (ie. not publicly advertised) fields which have been provided to me |
| 19:11.28 | dan_j | The* |
| 19:13.08 | Samot | What model is it? |
| 19:13.13 | dan_j | KX-UT133 |
| 19:13.15 | dan_j | Panasonic |
| 19:13.38 | dan_j | So, assuming TLS is not available, am I correct that there is no way to encrypt the SIP call data? |
| 19:13.50 | Samot | Nope not one that model. |
| 19:14.03 | Samot | Does the KX-UT support TLS? Yes, depending on the model. |
| 19:14.09 | Samot | Entry level models, no. |
| 19:14.13 | Samot | Higher level models, yes. |
| 19:14.42 | Samot | Googling: Panasonic KX-UT TLS support found me those answers in a few clicks. |
| 19:16.01 | Samot | The 123/133 models are the _only_ two that don't support TLS. |
| 19:16.05 | dan_j | Samot: I cant find it on any google results. even for the higher models |
| 19:16.10 | dan_j | I'll check the higher models |
| 19:16.27 | Samot | 200, 300, 600 all support TLS according the tech specs I have found. |
| 19:17.20 | dan_j | there are no kx-ut300 models, so you are probably looking at their newer version kx-hdv |
| 19:17.43 | [TK]D-Fender | dan_j> So, assuming TLS is not available, am I correct that there is no way to encrypt the SIP call data? <- does it say there is a VPN option built in? |
| 19:17.57 | Samot | KX-UTG. |
| 19:17.58 | dan_j | no vpn built in |
| 19:18.16 | [TK]D-Fender | Then why are you asking questions you already know the answer for? |
| 19:18.26 | [TK]D-Fender | the protol isn't secure. It can't tunnel it |
| 19:18.32 | [TK]D-Fender | that IS the answer |
| 19:18.37 | [TK]D-Fender | protocol* |
| 19:19.01 | [TK]D-Fender | tls secures sip. |
| 19:19.16 | [TK]D-Fender | vpn secures any packets |
| 19:19.24 | [TK]D-Fender | If you don't have one then you have NONE |
| 19:19.40 | dan_j | [TK]D-Fender: I just wanted to clarify that this model phone is inherently insecure when used over the internet. Thanks |
| 19:20.00 | Samot | That depends. |
| 19:20.14 | Samot | It supports HTTPS provisioning/applications.. |
| 19:20.39 | dan_j | Yep |
| 19:20.45 | Samot | So no, it's not "inherently insecure" |
| 19:20.54 | dan_j | Bit pointless if someone can crack the MD5 password from the sip packets |
| 19:21.05 | Samot | ? |
| 19:21.20 | Samot | You think TLS makes your call 100% secure? |
| 19:21.36 | dan_j | Nothing is 100% secure nowadays |
| 19:22.09 | Samot | You're under the assumption is ever was. |
| 19:22.22 | Samot | IP security has only gotten better over the years. |
| 19:23.20 | Samot | Bottom line: These phones do not do what you want. |
| 19:23.34 | dan_j | Ok, it never was, but with the speed of computing power, its easier to break than ever |
| 19:24.52 | Samot | Are you looking to secure the data being sent to the phone such as provisioning data and accessing the GUI.. |
| 19:24.58 | Samot | Or are you looking to secure your calls? |
| 19:25.53 | [TK]D-Fender | https://jive.com/includes/downloads/phones/panasonic-1XX-admin-guide.pdf |
| 19:26.10 | [TK]D-Fender | No encryption other than to the provisioning files themselves |
| 19:27.16 | dan_j | Samot: I've always secured the provisioning data and disabled the GUI. I changed the SIP password on my asterisk platform but did not send it to the client immediately. 24 hours later, I set the password on the phone, and within a few hours, the hacker was able to make calls. |
| 19:27.31 | Samot | From the same IP address? |
| 19:27.34 | Samot | Or IP block? |
| 19:27.47 | dan_j | Samot: Same IP, but I've now blocked the whole country. |
| 19:28.06 | dan_j | Although its not hard for the hacker to buy a new ip from various VPN providers. |
| 19:28.11 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-wkkgxnwiifcjxkxm) |
| 19:28.23 | Samot | Stop making excuses. |
| 19:28.37 | Samot | These are part of security risk mitigation. |
| 19:28.45 | WIMPy | Oh. That thing. |
| 19:29.14 | WIMPy | How do you send the password to the phone? |
| 19:29.24 | Samot | ^ Beat me to it. |
| 19:29.56 | dan_j | Samot: I have mitigation in place to prevent large amounts of fraudulent dialling, but there is currently an asterisk bug allowing hundreds of calls to be placed above the call-limit so the fraud system doesnt kick in until the calls have ended and the CDRs updated. |
| 19:30.18 | Samot | So you mitigated being charged money. |
| 19:30.27 | dan_j | WIMPy: I used the encrypted provisioning files. My next step is to manually config the phone using the GUI and then turn off GUI access. |
| 19:30.35 | Samot | You haven't mitigated on stopping people from getting in. |
| 19:30.50 | WIMPy | What version is broken regarding call limits? |
| 19:31.03 | dan_j | Seems to be all currently. 1sec. |
| 19:31.16 | dan_j | https://issues.asterisk.org/jira/browse/ASTERISK-26496 |
| 19:31.18 | WIMPy | And you can always use the GROUP function to do it yourself which has officially been the way to go for quite some time. |
| 19:31.34 | dan_j | WIMPy: GROUP is broken. See above. |
| 19:31.57 | WIMPy | BOTH are broken? |
| 19:32.04 | dan_j | Yep. |
| 19:32.26 | WIMPy | Wow. |
| 19:33.01 | dan_j | I'm not sure if the same occurs with v13 and PJSIP. |
| 19:33.38 | Samot | [2016-10-22 09:00:15] VERBOSE[14436][C-002fa42b] pbx.c: -- Executing [1556986479@subDialTrunk:4] Set("Local/1556986479@mora-temprana-000e7be0;2", "MAXCHAN=100") in new stack |
| 19:33.45 | WIMPy | Ok, that doesn't look that bad. |
| 19:33.57 | Samot | [2016-10-22 09:00:15] VERBOSE[14437][C-002fa42a] pbx.c: -- Executing [1555899130@subDialTrunk:6] Set("Local/1555899130@mora-temprana-000e7bdf;2", "GCOUNT=32") in new stack |
| 19:34.16 | WIMPy | It's "just" a race condition when calls are received simultaneousely. |
| 19:34.24 | Samot | Why are they executing Local channels? |
| 19:34.30 | dan_j | WIMPy: thats what the hacker is currently doing. |
| 19:34.37 | dan_j | Samot: they are using local channels as an example |
| 19:34.49 | Samot | Who is "they"? |
| 19:34.50 | WIMPy | But you should be able to circumvent that by checking the GROUP_COUNT later or twice. |
| 19:34.57 | Samot | Is that debug from Rusty's test? |
| 19:35.19 | dan_j | Samot: Rusty has provided that example for the developer. It occurs with standard channels too |
| 19:35.25 | Samot | Where's a debug from _your_ hacks? |
| 19:35.37 | dan_j | Samot: that is not my ticket |
| 19:35.51 | Samot | OK, doesn't change the answer. |
| 19:36.02 | Samot | Show how it's not working. |
| 19:36.52 | Samot | because in this debug I see MAXCHAN=100 and GCOUNT=32 |
| 19:37.03 | Samot | I'm trying to figure out how that's not working.. |
| 19:37.41 | dan_j | Surely the first GCOUNT should be 32 and the GCOUNT on the next line should say 33 ? |
| 19:38.24 | Samot | Unless like WImPy said if two calls happen at the same time, they might get the same count. |
| 19:38.47 | dan_j | Using Rusty's example dialplan, GROUP_COUNT always returns 1, rather than 2 for the 2nd call. |
| 19:39.05 | Samot | So you've setup an extension.. |
| 19:39.11 | WIMPy | Actually on Rustys example the behaviour seems to be correct. |
| 19:39.19 | Samot | Set it's OB limit to 2 |
| 19:39.29 | dan_j | OB? |
| 19:39.36 | Samot | Made a call, put it on hold, made another call, put it on hold and made a third call? |
| 19:39.43 | WIMPy | Yes, but he reads the count before setting the group. So that looks correct to me. |
| 19:39.48 | Samot | Outbound |
| 19:40.05 | dan_j | Samot: that works, because its not simultaneous |
| 19:40.34 | dan_j | WIMPy: his dialplan is wrong then. I'll test it. |
| 19:40.43 | Samot | How many simultaneous calls are they sending? |
| 19:41.03 | WIMPy | And I just tried to set call-limit and it works for me. |
| 19:41.21 | WIMPy | 11.24.1-w1 |
| 19:41.32 | Samot | Well we just went from it never working to only when it's simultaneous calls. |
| 19:41.48 | WIMPy | that was several minutes ago :-) |
| 19:41.54 | Samot | So it's not "completely" broken. |
| 19:42.15 | WIMPy | I haven't got any proof that it's broken at all, yet. |
| 19:42.26 | Samot | dan_j: How many simultaneous calls are they sending? |
| 19:42.40 | dan_j | about 15, when call-limit is set to 2 |
| 19:43.00 | Samot | Do you have a debug of this? |
| 19:43.05 | Samot | Or something from the full log? |
| 19:43.09 | WIMPy | And they are definitely all using the same account? |
| 19:44.30 | Samot | Is the phone behind NAT and on DHCP? |
| 19:44.46 | dan_j | Samot: I've got the verbose CLI from when it happened. 1sec |
| 19:44.55 | WIMPy | Are your call-counters working at all? |
| 19:45.05 | dan_j | WIMPy: You are correct. Rusty's code was wrong and it seems to work fine with local channels. |
| 19:45.50 | dan_j | WIMPy: yes, because when I make non-simultaneous calls, the limit is honoured. |
| 19:45.56 | Samot | Is the phone behind NAT and on DHCP? |
| 19:46.14 | WIMPy | A rate limit might help then. |
| 19:46.19 | dan_j | The phone is behind a NAT and on DHCP. |
| 19:47.36 | WIMPy | But just using the GROUP functions as I suggested earlier should do the trick, it seems. |
| 19:48.25 | Samot | Let's see what is happening in the debug. |
| 19:53.51 | dan_j | http://pastebin.com/c2zec3jm |
| 19:54.31 | dan_j | Priority 8 and 9 are: |
| 19:54.35 | dan_j | exten => _9.,n,NoOp(CallCountForPeer = ${GROUP_COUNT(1@OutgoingCallsForPeer${DIALLINGPEER})}) |
| 19:54.35 | dan_j | exten => _9.,n,GotoIf($[${GROUP_COUNT(1@OutgoingCallsForPeer${DIALLINGPEER})}>${ODBC_PEERINFO(${DIALLINGPEER},maxconcurrentoutgoing)}]?toomanycalls) |
| 19:54.50 | dan_j | Sorry, that first line is new |
| 19:55.08 | dan_j | Priority 8 is exten => _9.,n,Set(GROUP(OutgoingCallsForPeer${DIALLINGPEER})=1) |
| 19:55.35 | dan_j | This code works for non-simultaneous calls. |
| 19:56.05 | dan_j | But should be irrelevant when the call-limit is lower than the maxconcurrentoutgoing |
| 19:57.08 | dan_j | In the above case, call-limit is set to 2. |
| 19:57.29 | WIMPy | It's dirty, but I guess a Wait(0.1) in between would fix it. |
| 19:57.35 | *** join/#asterisk lacy (~lacy@74.95.148.242) |
| 19:58.06 | dan_j | Just wondering how I can test it. Probably need to set up another asterisk server in order to fire INVITEs at it. |
| 19:58.17 | dan_j | Or a couple of phones |
| 19:59.22 | WIMPy | sipsak? |
| 20:00.08 | WIMPy | But make sure you can reliably make it fail before you change things. |
| 20:04.28 | Samot | What's with all the DTMF? |
| 20:06.19 | Samot | Priority 8 is exten => _9.,n,Set(GROUP(OutgoingCallsForPeer${DIALLINGPEER})=1) |
| 20:06.29 | Samot | ^ Why are you always setting the call count to 1? |
| 20:07.03 | Samot | If that is your current code, you are always telling it "Hey I got one call" |
| 20:07.32 | dan_j | Samot: I assume the hacker is using the DTMF to check whether he has been successful. Got something listening for the DTMF on the other side? |
| 20:09.25 | Samot | OK. |
| 20:09.33 | dan_j | Samot: I've putting that channel into that group. Even when I use that code, GROUP_COUNT is always the number of calls in that group, regardless of whether i use =1 |
| 20:09.45 | dan_j | Unless I've missed something. |
| 20:09.48 | dan_j | But it works |
| 20:10.18 | WIMPy | Looks ok to me. Just a group called "1". |
| 20:12.17 | Samot | Did you post your dialplan? |
| 20:17.02 | dan_j | No? Is it required? |
| 20:19.43 | WIMPy | So what's going on? Do you have a test case? |
| 20:19.56 | WIMPy | Or do youneed more solutions to stop the issue? |
| 20:23.30 | dan_j | I'm just trying to get to grips with sipsak |
| 20:24.54 | WIMPy | good |
| 20:28.22 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 20:53.01 | dan_j | WIMPy: test finished. call-limit seemed to work on the first attempt, but GROUP was wrong. |
| 20:55.19 | dan_j | https://www.irccloud.com/pastebin/9hC7jhuN/ |
| 20:56.07 | dan_j | I'm going to try the test again but with the peer that is being hacked. |
| 21:01.36 | dan_j | Nope. My tests are all blocked by call-limit which is set to 2. I think it's because I've not worked out how to get sipsak to send simultaneous invites. I'm running sipsak multiple times using a bash script, but it must be delayed just enough to allow call-limit to be calculated. |
| 21:02.59 | dan_j | GROUP works fine when I run each INVITE manually, but when they are grouped in the bash file, the GROUP_COUNT does not increase. |
| 21:03.10 | dan_j | Trying with Wait(0.5) now. |
| 21:03.55 | dan_j | Wait didn't help. Still the same result. |
| 21:16.21 | WIMPy | Hmm. that's interesting. |
| 21:17.59 | dan_j | I was thinking of putting into place my own 'GROUP' by using shared variables. until the issue is fixed. |
| 21:18.12 | WIMPy | You put the Wait between the Set(GROUP and the GROUP_COUNT? |
| 21:18.41 | WIMPy | IF it gets fixed. |
| 21:19.29 | WIMPy | You might know my standard answer to most questions: Listen on AMI. You can always count and kill calls from there. |
| 21:19.37 | dan_j | If it doesnt then I'll have to set up an asterisk box and try to find which version broke the count |
| 21:20.13 | dan_j | ...if it ever worked |
| 21:21.07 | dan_j | 21:18Â <WIMPy>Â IF it gets fixed. <<< I would hope that something most people use to mitigate fraud would be fixed fairly quickly. |
| 21:23.48 | WIMPy | Hope is good for your endurance but blurrs your judgement. |
| 21:27.24 | Samot | Well I have things like blocking unwanted IP blocks from known bad locations so they can't access the network. |
| 21:28.15 | Samot | Block high cost destinations. |
| 21:28.19 | Samot | Limit NANP calling. |
| 21:29.00 | dan_j | I'm about to remove dialling to countries that I know my clients don't dial to. |
| 21:30.28 | Samot | I also don't allow blanketed International dialing. |
| 21:30.29 | Samot | It has to be requested and even then it's broken out into three levels. |
| 21:32.10 | Samot | But I also don't do any billing or routing/switching processes on Asterisk servers. |
| 21:32.40 | dan_j | I dont do billing on the asterisk server. |
| 21:32.50 | Samot | But I don't check rates. |
| 21:32.57 | Samot | I don't do any of that on Asterisk. |
| 21:33.03 | Samot | I have Kamailio for that. |
| 21:33.08 | dan_j | I check rates as some clients want high rate destinations. |
| 21:33.35 | Samot | I'm just saying, I don't do that on Asterisk. |
| 21:33.42 | Samot | I use Kamailio for that. |
| 21:33.54 | Samot | Clients have to go through Kamailio before getting to Asterisk.. |
| 21:34.03 | dan_j | ok |
| 21:34.27 | Samot | I also have Kamailio servers between Asterisk and PSTN.. |
| 21:35.15 | Samot | To manage incoming routes from carriers and do all the rate checking, channel limits along with doing it for outbound.. |
| 21:35.18 | Samot | Including LCR. |
| 21:38.14 | Samot | Outside of security it gives me one central location for all the PBX systems I have |
| 21:38.48 | Samot | So I'm not dealing with the same script running on multiple boxes. |
| 21:39.12 | Samot | Eases the amount of work Asterisk has to do.. |
| 21:39.42 | dan_j | Samot: The plan to block the entire country didnt help. Just looking at the asterisk logs. The hacker registered from a UK IP and tried to dial yesterday after I blocked the Palestinian territories. |
| 21:40.41 | Samot | Is the device at a location with a static WAN? |
| 21:41.06 | dan_j | I've asked the client to get a static IP from his ISP and then i'm going to block everything else on that peer. |