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| 02:01.45 | bobbyz | Hi guys. Anyone here used Flowroute? I'm wondering if they're a decent provider |
| 02:02.33 | Penguin | They are. |
| 02:02.55 | bobbyz | thanks, Penguin |
| 02:08.01 | bobbyz | I looked through the list of T38 providers on voip-info, and I was most attracted to Flowroute. I run a tiny IT consulting company in the US and have asterisk experience. I was considering dropping my land lines for a SIP provider that supported T38. Any other small-business and US-friendly SIP w/T38 providers stick out? |
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| 04:54.32 | MaliutaLap | .j #sage-au |
| 04:54.36 | MaliutaLap | oops |
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| 06:41.46 | justdave | is there a session variable that'll have the remote IP address of an incoming anonymous SIP call? |
| 06:42.06 | justdave | er, channel variable |
| 06:43.33 | justdave | as usualy, I find it after I ask. looks like SIP_RECVADDR |
| 06:43.45 | justdave | except it's only available on Asterisk 11 and later, and I'm on 1.8 still :( |
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| 06:45.26 | justdave | we allow anonymous inbound calls from the internet, and have it well locked-down so they can only call a set of extension numbers, can't make outbound calls, etc. Got someone repeatedly attempting to make outbound calls, so I was hoping to figure out how to get the IP address so I could make fail2ban block them |
| 06:45.54 | justdave | (they all fail, but if you can tell they're making mischief, no point in letting them make the attempt) |
| 06:50.34 | ChannelZ | Look at the CHANNEL function |
| 06:50.51 | ChannelZ | peerip, recvip |
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| 06:52.14 | justdave | aha, thanks! |
| 06:53.43 | ChannelZ | although doesn't it already say the IP for 'extension not found' notices? |
| 06:53.54 | ChannelZ | (I don't remember for 1.8) |
| 06:54.32 | justdave | just the SIP channel name |
| 06:54.37 | ChannelZ | Like: Call from '' (1.2.3.4:5060) to extension '1111' rejected because extension not found...." |
| 06:54.44 | justdave | which has an IP address in it, but it's my system's IP address not the remote one |
| 06:55.04 | ChannelZ | Those are probably registrations |
| 06:55.23 | justdave | yeah registration failures have that in them |
| 06:55.34 | ChannelZ | I remember hacking the source to make it output the remote IP on one of those logs, I just can't remember which one. |
| 06:57.56 | justdave | ok, got a NoOp with those in the log in the context that handles anonymous calls now, let's see what it tells me. |
| 06:59.38 | ChannelZ | if you need me to try let me know |
| 06:59.45 | justdave | we've got SRV records for our domain set up so you can dial by a sip: URI that is just the user's email address. It shells to an AGI that does a database lookup to match the email to an extension number, but falls back on attempting extension numbers if it doesn't match anything. |
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| 07:00.22 | justdave | so I'm getting lots of invalid extension with attempts to dial numbers in Israel |
| 07:00.48 | ChannelZ | Mine are always palestine |
| 07:02.57 | justdave | ok, just had one of those on another server... |
| 07:03.13 | justdave | [Nov 15 23:02:18] NOTICE[2345]: chan_sip.c:22650 handle_request_invite: Call from '' (178.162.205.207:5071) to extension '9011972592770237' rejected because extension not found in context 'from-invalid-sip'. |
| 07:03.38 | ChannelZ | yeah so that IP should be the correct remote |
| 07:03.38 | justdave | that one's not set up to do the dial-by-email thing (it's not the target of the SRV records) |
| 07:03.52 | justdave | the dial-by-email doesn't get that message becaue it has a catchall dialplan |
| 07:04.04 | justdave | so anything that comes in matches |
| 07:04.08 | justdave | and then gets rejected later |
| 07:04.27 | ChannelZ | ah |
| 07:05.51 | justdave | there we go, got one |
| 07:05.52 | justdave | <PROTECTED> |
| 07:06.02 | justdave | whadayaknow, it's the same damn IP address hitting both servers |
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| 07:07.51 | ChannelZ | Yeah I see that all the time |
| 07:08.29 | ChannelZ | In fact scrolling back I had someone in that same Class C earlier.. |
| 07:08.38 | ChannelZ | 178.162.205.231 and 178.162.205.199 |
| 07:09.33 | justdave | that block appears to be in Germany |
| 07:09.47 | ChannelZ | Leaseweb slum |
| 07:10.59 | justdave | ok, now what I need to figure out how to do is to proactively look up in the dialplan if that extension exists or not, and log something fail2ban can look for it if doesn't. |
| 07:11.40 | justdave | invalid handler in the extensions context would be the obvious place, but I don't want it there because that would flag legitimate people that just misdialed |
| 07:11.45 | justdave | only want it on the anonymous calls |
| 07:12.30 | ChannelZ | They don't come in a separate context? |
| 07:13.21 | justdave | they do |
| 07:13.37 | justdave | but it jumps to the number they used in the extensions context if the email lookup fails |
| 07:14.02 | ChannelZ | and the repetition/timeout of fail2ban should help figure out if it's someone who misdialed once or is trying multiple times in a short period to dial some external number |
| 07:14.12 | justdave | DIALPLAN_EXISTS() looks like what I want here. |
| 07:14.16 | ChannelZ | I go with "3 strikes and you're out" |
| 07:15.36 | justdave | peerip - R/O Get the IP address of the peer. |
| 07:15.37 | justdave | recvip - R/O Get the source IP address of the peer. |
| 07:15.41 | justdave | what's the difference? :) |
| 07:16.33 | ChannelZ | One might refer to a source through a proxy, I don't know |
| 07:17.29 | ChannelZ | like recvip is probably the IP the packet came from, but that could be a SIP proxy and the actual peer calling is peerip (guessing) |
| 07:18.10 | Penguin | I think of peers that are being NAT. |
| 07:18.33 | Penguin | The packets might have a source of one address and a different address in the packet. |
| 07:18.41 | ChannelZ | that could be it |
| 07:18.58 | Penguin | I'd test both fields to see what data is in them. |
| 07:20.33 | Penguin | I just got here, so I didn't know what you guys were talking about.. |
| 07:20.55 | Penguin | . but it looks like you're talking about catching host addresses to ban them. |
| 07:21.10 | ChannelZ | Indeed |
| 07:21.39 | Penguin | I use Log() and ${CHANNEL(recvip)} with great success. |
| 07:22.03 | justdave | I'm behind NAT and when I call it from here I get the same IP address in both |
| 07:22.20 | Penguin | I was thinking of nat on the far end. |
| 07:22.42 | justdave | so am I |
| 07:22.47 | justdave | (I'm not in the same place with the phone server) |
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| 07:23.24 | Penguin | I had it in my mind that peerip could show something like 192.168.1.43, where recvip might be 74.161.45.2. |
| 07:23.40 | Penguin | I would want to block the recvip, not the peerip. |
| 07:28.07 | Penguin | If you have the catch-all extension configured that all the dialing attempts will hit, you can use Log(NOTICE,${CHANNEL(recvip)} is attempting to make unauthorized calls); and then in fail2ban's filter.d/asterisk.conf regex, you can use something like NOTICE.* .*: <HOST> is attempting to make unauthorized calls |
| 07:28.23 | Penguin | It's not exactly what I use, but that should give you an idea. |
| 07:48.02 | justdave | is the traditional dialplan language completely gone in favor of AEL in Asterisk 11? |
| 07:48.24 | Penguin | What would make you suggest such a thing? |
| 07:48.35 | justdave | no documentation on the wiki? |
| 07:48.41 | justdave | only docs for AEL |
| 07:48.53 | justdave | https://wiki.asterisk.org/wiki/display/AST/Configuration+and+Operation |
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| 07:50.11 | ChannelZ | The sample is pretty much the documentation |
| 07:51.35 | ChannelZ | But you can start here: https://wiki.asterisk.org/wiki/display/AST/Dialplan+Fundamentals |
| 07:52.08 | ChannelZ | (the Wiki as documentation is kind of poorly organized) |
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| 08:00.45 | justdave | where do I find expression syntax? |
| 08:01.49 | justdave | exten => _.,n,GotoIf($[DIALPLAN_EXISTS(local-extensions,${dialexten},1)]?local-extensions,${dialexten},1:notlocal) |
| 08:01.54 | justdave | that gives me |
| 08:01.55 | justdave | [Nov 15 23:54:35] WARNING[25867]: ast_expr2.y:1333 op_minus: non-numeric argument |
| 08:02.28 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Expressions |
| 08:03.42 | ChannelZ | DIALPLAN_EXISTS needs ${} |
| 08:06.18 | justdave | aha, thanks |
| 08:11.22 | Penguin | Functions are treated like variables. |
| 08:12.40 | ChannelZ | (keep that in mind for the Set application..) |
| 08:13.18 | justdave | ok, looks like it all works now. |
| 08:17.00 | justdave | yep, and this works nicely for something I can trap with fail2ban |
| 08:17.01 | justdave | exten => _.,n,Log(NOTICE,"Anonymous call from ${CHANNEL(recvip)} sent to invalid extension ${dialexten}") |
| 08:17.13 | Penguin | I wouldn't use that pattern. |
| 08:17.32 | Penguin | _X. would be better. |
| 08:17.45 | justdave | X requires it start with a number |
| 08:17.50 | justdave | I'm explicitly looking for text |
| 08:17.59 | Penguin | I still wouldn't use _. as the pattern. |
| 08:18.12 | Penguin | If you need letters, form a pattern that has letters. |
| 08:18.17 | justdave | ok, how do I make that match anything alphanumeric? |
| 08:19.02 | Penguin | Keep in mind, most of the illegitimate dial attempts aren't going to be sending letters. |
| 08:19.44 | Penguin | And where did you come up with this dialexten variable? |
| 08:19.50 | justdave | right, but legitimate ones will be (and this is in the same extension after it determines that what they requested was invalid) |
| 08:19.59 | justdave | the dialexten variable is returned from an AGI script |
| 08:20.26 | justdave | which returns the matching extension number if the email address that got passed in is found in the database |
| 08:20.39 | justdave | and the actual string they passed in untouched if it wasn't |
| 08:21.16 | justdave | this is in the anonymous call handler for DNS SRV records for our email domain |
| 08:21.16 | Penguin | Why not do something with the bad calls before it ever reaches that point? |
| 08:21.33 | justdave | how do I know they're bad before they get to that point? |
| 08:22.04 | Penguin | If you are only allowing dial by email address, the fact that it is a number would mean it's a bad call. |
| 08:22.27 | justdave | actually allowing extension numbers as well, if they exist |
| 08:22.39 | Penguin | Okay, that makes it different. |
| 08:23.55 | Penguin | So now you want to know a pattern, but the pattern needs to be alphanumeric... |
| 08:24.54 | Penguin | I'd have to test for accuracy, but _[a-z0-9]. would be what I test. |
| 08:25.34 | Penguin | The reason for that is so that your pattern does not match the asterisk standard extensions. |
| 08:26.11 | Penguin | You don't want your pattern to match o, s, h, i, t, a, fax. |
| 08:26.51 | Penguin | (fax would be okay, since fax is produced by faxdetection) |
| 08:36.55 | justdave | cool, that seems to work |
| 08:37.20 | justdave | and got rid of all the warnings in the log on dialplan reload about "you really don't want to use _." too :) |
| 08:37.41 | Penguin | You mean you got that warning from asterisk and you still used it? |
| 08:38.29 | justdave | didn't know you could do sets in the pattern match, that helps. |
| 08:38.54 | justdave | (well, set ranges, I knew you could do individual numbers) |
| 08:39.05 | justdave | didn't want to list the entire alphabet :) |
| 08:39.46 | Penguin | I'm pretty sure you can even do things like [a-cf-ix-z] |
| 08:40.12 | Penguin | Maybe you want to skip some stuff. |
| 08:42.15 | Penguin | Anyway, that will give you something to think about for a while. I'm heading to bed. |
| 08:42.19 | Penguin | Good luck. |
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| 08:46.03 | justdave | yeah, me too, it all works. Thanks for the help! |
| 08:56.21 | ChannelZ | O shit a fax! |
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| 11:04.30 | [sr] | hi |
| 11:04.49 | [sr] | who can give me a hand creating a speed dial by hand ? |
| 11:04.50 | [sr] | :p |
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| 11:05.14 | [sr] | i could use: exten => 159,1,Dial(DAHDI/g1/961111111) |
| 11:05.21 | [sr] | but no cdr, call recording, etc etc |
| 11:06.19 | kaldemar | core show application Dial |
| 11:06.42 | kaldemar | you need to specifically disable cdr to not get a record for a call. |
| 11:10.01 | [sr] | in my case i want to enable it |
| 11:10.06 | [sr] | not disable it |
| 11:10.25 | [sr] | let me read a bit |
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| 14:15.49 | ghost75 | is it possible that conference softbutton on cisco phone is not working because it sends some proprietary "x-cisco...." something to * ? |
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| 16:02.18 | Penguin | ghost75: Probably not. |
| 16:02.52 | ghost75 | x-cisco-remote-cc i saw as ip paket |
| 16:03.41 | Penguin | In sip debug? |
| 16:03.57 | ghost75 | wireshark |
| 16:04.05 | ghost75 | in sip |
| 16:04.22 | Penguin | Let's worry about what asterisk is doing with the packets. |
| 16:05.32 | ghost75 | denying |
| 16:05.54 | Penguin | Denying what? |
| 16:06.16 | Penguin | I'm not seeing the debug. |
| 16:06.18 | ghost75 | i dont know, i think was response |
| 16:06.33 | ghost75 | i did 2 days ago |
| 16:06.50 | Penguin | We're talking about it now. |
| 16:07.47 | ghost75 | this is what somebody writes: 1. Conference. When trying to join 2 current calls into a 3-way conference using the conference button, the phone displays "Unable to complete conference". The Asterisk CLI (verbosity 3) reports "WARNING[11481]: chan_sip.c:23126 handle_refer_remotecc: Unknown softkeyevent: Conference" |
| 16:07.51 | ghost75 | https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 16:07.53 | LieutPants | [ASTERISK-13145] [Status: Open] [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML - https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 16:08.47 | Penguin | What is the phone model? |
| 16:08.59 | ghost75 | 7941 |
| 16:09.52 | ghost75 | guess i need a newer patch for * and cisco |
| 16:16.02 | [TK]D-Fender | Cisco's phones seem lazy & dumb |
| 16:16.24 | [TK]D-Fender | They somehow expect the server to do the 3-way call work for them. |
| 16:16.30 | [TK]D-Fender | This isn't SCCP |
| 16:16.34 | [TK]D-Fender | or MGCP |
| 16:16.51 | [TK]D-Fender | Polycom takes this stuff for granted.. |
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| 16:43.59 | apb1963 | So... here we are...20 minutes before my conference starts.... and I get a busy signal using asterisk/Google Voice.... but my landline has no problem getting through. :( |
| 16:44.26 | *** join/#asterisk davlefouAMD (~david@41.225.216.74) |
| 16:44.44 | Penguin | Dial it again. |
| 16:45.12 | Penguin | I might dial it, too, just to check. |
| 16:45.28 | apb1963 | same. |
| 16:45.30 | apb1963 | go ahead |
| 16:45.46 | apb1963 | need the number or still have it? |
| 16:45.56 | Penguin | I was going to look it up in cdr. |
| 16:46.08 | Penguin | 805 number? |
| 16:46.11 | apb1963 | yes |
| 16:46.14 | Penguin | Got it right here. |
| 16:46.24 | apb1963 | go for it |
| 16:47.56 | Penguin | I tried two times and got the busy tone both times. |
| 16:48.17 | apb1963 | hmm |
| 16:49.46 | apb1963 | I still can't wrap my head around the problem... you and I can't both be taking the same route can we? |
| 16:50.13 | Penguin | We most certainly can. The route is between Google and that conference. |
| 16:50.22 | Penguin | But it was a regular busy tone, not congestion. |
| 16:50.36 | apb1963 | Which means? |
| 16:50.46 | Penguin | It means the far end is in use. |
| 16:50.57 | apb1963 | And yet... I can get through on my landline |
| 16:51.13 | Penguin | Try both paths a few more times. |
| 16:51.30 | Penguin | If you get busy on your landline just once, will that make you feel better? |
| 16:51.33 | apb1963 | landline good... |
| 16:51.44 | apb1963 | lol... doubtful |
| 16:52.15 | apb1963 | GV bad |
| 16:58.48 | file | GV will be GV. |
| 17:01.11 | apb1963 | yet typically after the conference is over... I can get through :( |
| 17:01.31 | apb1963 | Here's an interesting test... I'm gonna try through their webpage |
| 17:05.40 | apb1963 | It seems to be ringing... no busy signal... but I'm not sure it picked up or not...I don't think it did |
| 17:05.59 | apb1963 | but the ringin stopped |
| 17:07.45 | apb1963 | ok, 4 rings and then it comes back ready for another call |
| 17:08.00 | apb1963 | so it doesn't pick up |
| 17:11.49 | Penguin | I tried a few times between my regular route and via GV, and GV always goes busy while the other make it to the conf service. |
| 17:12.20 | Penguin | Very strange. |
| 17:14.41 | *** join/#asterisk serafie1 (~erin@24.96.64.240) |
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| 17:21.12 | *** part/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:23.12 | apb1963 | I agree |
| 17:31.42 | jeev | how do i get a dahdi debug ? |
| 17:31.47 | jeev | pri set debug on isn't working |
| 17:32.14 | jeev | pri show channels lists fine |
| 17:32.51 | jeev | span 1 i guess |
| 17:33.10 | jeev | localhost*CLI> pri set debug on span 1 |
| 17:33.10 | jeev | localhost*CLI> |
| 17:33.19 | jeev | i can't do anything after that, jsut takes commands and does not process |
| 17:33.43 | jeev | arigt now it's ok |
| 17:36.03 | jeev | PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established) <- yay |
| 17:36.11 | jeev | not sure but looks like an error. |
| 17:36.24 | *** join/#asterisk mitchrodrigues (~mitchrodr@184.250.150.22) |
| 17:37.51 | Penguin | It does say error. |
| 17:38.30 | jeev | any idea? telco can't figure i tout. |
| 17:50.16 | WIMPy | Might be some one-way communication. |
| 17:51.52 | *** join/#asterisk Sheeplet (~chiron@197.87.54.233) |
| 17:51.59 | Sheeplet | lo |
| 17:54.07 | jeev | they say my d-chan isn't up but it is. |
| 17:54.35 | WIMPy | They can't receive you. |
| 17:54.42 | WIMPy | Have you tried another cable? |
| 17:57.01 | jeev | no, i'm not on site |
| 17:57.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.57) |
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| 18:03.29 | *** join/#asterisk IamTrying (~IamTrying@82.143.92.22) |
| 18:04.12 | IamTrying | If i use fail2ban for FreeSwitch and Asterisk where fail2ban has 5060, 5061, 5080, 5081 which one it will look for? Asterisk or FreeSwitch? |
| 18:05.40 | [TK]D-Fender | It doesn't monitor port.. it reads logs |
| 18:05.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:05.48 | [TK]D-Fender | so it'll monitor whatever YOU tell it to look at |
| 18:08.15 | IamTrying | [TK]D-Fender, OK - i ran Asterisk and fail2ban (and necessary Iptables) but users cant register smoothly anymore from 3G networks. And when i terminate the call the PDD is huge high like 4 minute to terminate call. What specifically should be the IPtables for B-leg? |
| 18:08.31 | ghost75 | [TK]D-Fender: not sure if cisco planned that sip fw to be used against non cisco pbx oO |
| 18:09.08 | [TK]D-Fender | IamTrying: Too vague... |
| 18:09.27 | [TK]D-Fender | ghost75: And Cisco PBX's speak SCCP so there'd be no reason to use SIP at all |
| 18:09.51 | ghost75 | yeah thats i where i start thinking why they did sip fw at all |
| 18:09.58 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 18:10.17 | [TK]D-Fender | not a great choice for Asterisk... |
| 18:10.47 | IamTrying | [TK]D-Fender, Registration is OK, but the call termination fails, is it a Asterisk Bug with fail2ban? |
| 18:11.05 | [TK]D-Fender | "termination fails" is also vague |
| 18:11.21 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 18:12.05 | Penguin | Asterisk and fail2ban are not related, so tehre is no asterisk bug for fail2ban. |
| 18:12.22 | Penguin | Fail2ban reads log files and matches patterns. |
| 18:12.25 | Penguin | That is all. |
| 18:14.11 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 18:17.50 | ghost75 | is it iptables based? |
| 18:18.11 | IamTrying | Penguin, Then what else could cause the call termination to fail like after 1/2/3 minutes later it gives ring tone. When i switch off Asterisk, fail2ban, iptables and use FreeSwitch the same call is within 1-2 seconds ringing. |
| 18:18.18 | IamTrying | ghost75, YES iptables |
| 18:18.51 | Penguin | fail2ban does work with other firewalls, but it's usually used with iptables when on Linux. |
| 18:18.52 | [TK]D-Fender | Where do we see you looking at the call itself? |
| 18:20.37 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 18:24.37 | [TK]D-Fender | Fail2ban has nothing to do with it if the call is in progress |
| 18:26.48 | IamTrying | OK - Thank you [TK]D-Fender Penguin |
| 18:26.57 | ChannelZ | "when I change 4 separate things, it works!" |
| 18:27.20 | Penguin | Exactly. |
| 18:28.09 | outtolunc | usually takes me 5 :( |
| 18:28.27 | Penguin | You're still new. |
| 18:28.38 | outtolunc | yeah, <- newb |
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