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06:06.04 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.2 (2019/11/21) 16.6.2 (2019/11/21) Standard: 17.0.1 (2019/11/21); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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07:59.30 | wyoung | drmessano: so what's expensive? |
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11:19.04 | admin0 | for using asterisk wtih mysql/postgres, is there a page which lists the table i need to create in the db ? or will asterisk create it itself ? |
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12:20.37 | TandyUK | how do i verify that a tls cert is correctly installedi n asterisk... ive used https://www.geocerts.com/ssl-checker among other to validate the one on our web portal is correct, but not sure how to check the one asterisk is using is correct |
12:20.59 | TandyUK | we keep getting security warnigns when trying to log mobile apps into a sip account |
12:21.28 | TandyUK | trust the cert, and its fine, but there should be no need to do that if the cert/ca certs are done properly |
12:22.24 | sibiria | you could start with verifying that the certificate was issued by the CA, and that the cert and key match eachother |
12:22.42 | TandyUK | the cert itself is identical to the one we're using for https |
12:22.59 | sibiria | but if your clients reject the CA it's obviously a case of their CA storage not being adequate |
12:23.11 | TandyUK | i suspect somethign is wrong in the ca cert chain |
12:23.26 | sibiria | are you using lets encrypt? |
12:23.30 | TandyUK | no |
12:24.10 | TandyUK | https://www.geocerts.com/ssl-checker check "voip.tandyuk.com" port 5555 on there and you can see the full chain |
12:25.11 | sibiria | i'm sure it's fine |
12:25.20 | sibiria | but your clients need to have that CA in their store regardless |
12:26.01 | jkroon | TandyUK, openssl s_client -connect server:port </dev/null |
12:26.14 | jkroon | check that it's the cert you're expecting, including the chain. |
12:26.22 | sibiria | or just wget verbosely etc. |
12:26.31 | sibiria | or curl etc. |
12:26.45 | jkroon | indeed. |
12:27.06 | jkroon | wget + curl will fail if you're hitting SIP/TLS, but should be OK for SIP/WSS |
12:27.22 | jkroon | it will however fail with protocol errors, not cert errors :) |
12:27.25 | TandyUK | specifically its SIP/TLS that i need to verify |
12:27.48 | jkroon | you have your options above. |
12:28.10 | jkroon | i found that older phone dislike >> sha1 ... so on those you're pretty screwed. |
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12:36.57 | TandyUK | ok sorted cheers guys |
12:37.07 | TandyUK | it was using the wrong ca chain bundle |
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14:09.08 | Sibert | Hello everyone |
14:09.39 | Sibert | I'm having an issue with asterisk where it's seemingly limiting the outgoing 'To' field to 255 chars |
14:09.50 | Sibert | Does anyone know why that is and if I can get around that? |
14:13.06 | Corydon76 | Seems like it might be a hardcoded buffer limit. Why do you need more than 255 characters? |
14:14.17 | admin0 | for using asterisk wtih mysql/postgres, is there a page which lists the table i need to create in the db ? or will asterisk create it itself ? |
14:15.46 | Sibert | Corydon76, I'm using a proxy which supports push notifications (flexisip), which adds the necessary information to the contact |
14:15.56 | Sibert | This information is quite long |
14:17.10 | Samot | What should the To header look like? |
14:17.14 | Corydon76 | Sibert: you'll probably have to modify the length of the buffer in the code and recompile. |
14:17.40 | Corydon76 | Remember that it's going to be 256, since it needs the ending nul byte |
14:17.46 | Sibert | yeah oke |
14:17.46 | sibiria | Sibert: if this is with chan_sip, maybe try using pjsip to see if the limit is the same there |
14:17.55 | sibiria | (or browse the appropriate part of the pjsip source) |
14:18.10 | Sibert | I just found this in the output: |
14:18.35 | Sibert | sibiria, I'm not sure what it's using, it's a legacy project and I only know half what I'm doing |
14:18.40 | Corydon76 | admin0: You're going to need to be more specific. There are lots of ways that Asterisk can use a database. |
14:18.41 | Sibert | Is there an easy way to find out? |
14:18.48 | sibiria | i'd be betting that it's using chan_sip then |
14:19.03 | Samot | 9:17:11 AM <Samot> What should the To header look like? |
14:19.15 | Samot | There are limits. |
14:19.16 | sibiria | yes: inspect modules.conf to see what's being loaded, or run asterisk -x 'sip show peers' to confirm |
14:19.22 | Samot | tags can only be 255 characters. |
14:19.49 | admin0 | Corydon76, i just need one way that works fine .. where i can create sip accounts and the cdr is stored internally in the database |
14:19.53 | Sibert | Samot, you're saying there's no solution? |
14:20.02 | Samot | That's not what I said at all. |
14:20.11 | Samot | I asked to see what the To header should look like in full.. |
14:20.18 | Samot | So I could see it's formating |
14:20.20 | Corydon76 | admin0: those are two different tables and even two different connectors. |
14:20.22 | Samot | So I could see it's formatting |
14:20.22 | Sibert | Oh yeah give me a second, sorry |
14:20.46 | Corydon76 | admin0: I assume for the first, you're talking about SIP realtime? |
14:20.50 | Samot | How the tags are added, how many, how long are they, is the header properly formatted by the proxy |
14:21.30 | Sibert | sip:117@172.20.10.44:5061;app-id=<length 35>;pn-type=firebase;pn-timeout=0;pn-tok=<token with length 152> ;pn-silent=1;CtRt19224c02021ba6dc=tcp:172.20.10.43 |
14:21.37 | Sibert | this is the header that's being cut off |
14:22.16 | Sibert | the tags are added using the linphone sdk in the client |
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14:22.35 | Corydon76 | admin0: I don't know of a single case where Asterisk creates the table. The problem is that there are so many different databases, and while they are governed by a common standard, the creation of tables is something best left to an administrator. If you look in the contrib directory of the source, there should be some sample table creation scripts for some databases. |
14:23.22 | Sibert | sibiria, I don't seem to have a modules.conf |
14:23.25 | Samot | Well that header looks like. |
14:23.31 | Samot | er fine |
14:23.55 | Sibert | sibiria nvm found idk how I didn't see it |
14:24.03 | Samot | Yeah, I'd try doing this with PJSIP and make sure it's not a Chan_SIP limitation. |
14:24.36 | Sibert | okay, I'll try to figure that out |
14:24.39 | Sibert | thanks for the pointers |
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14:25.35 | Corydon76 | admin0: in the case of CDRs, I engineered the tables to be _extremely_ flexible. Asterisk scans the table structure and then matches CDRs elements and variables into the table layout. |
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14:26.46 | Corydon76 | Therefore, if you want to set a new CDR variable, something that doesn't exist anywhere else, and that field name exists in your CDR table, it will be logged, just as you might like. |
14:26.52 | Sibert | sibiria, neither modules.conf nor sip show peers seems to have a clear indication of what's being used |
14:29.22 | sibiria | if you get a positive response from "sip show peers" you are using chan_sip |
14:29.56 | sibiria | otherwise you'll get something along the lines of "no such command" or so |
14:30.26 | Sibert | okay thanks |
14:30.27 | sibiria | to speak to pjsip, try: asterisk -x 'pjsip show contacts' (and/or show endpoints for more details) |
14:30.33 | Sibert | Then I'm using chan_sip |
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14:34.33 | Sibert | Hm |
14:34.48 | Sibert | I'm running asterisk 11.13, which doesn't seem to have support for pjsip |
14:34.51 | Sibert | is that correct? |
14:37.37 | sibiria | no i think pjsip was available in 11. i think your build wasn't configured to include it |
14:37.49 | file | it was not |
14:38.05 | Sibert | Okay |
14:38.34 | sibiria | i recall using pjsip before ast 13 |
14:38.40 | sibiria | but maybe it was on 12 |
14:38.49 | seanbright | 12 was the first release with pjsip support |
14:39.20 | Sibert | Okay |
14:39.39 | Sibert | So in my current situation, I can't move forward unless I update asterisk |
14:40.28 | Samot | Well considering that means whatever version you are on is completely unsupported...... |
14:40.38 | Samot | Updating might be a wise choice. |
14:41.10 | Sibert | Yeah I suppose it is, but this is a inherited project full of dirty hacks, so I'm not sure if it's really an option |
14:41.45 | Samot | I'm not sure what a dirty hack would be in Asterisk. |
14:41.58 | Samot | Since there really isn't a default or standard setup for it. |
14:43.03 | Sibert | Well okay, to be fair, I'm not sure if it's dirty hacks, but web pages that directly write in the dialplan with hardcoded calls at least feel like dirty hacks |
14:43.51 | seanbright | ok |
14:44.02 | Samot | FreePBX is a GUI that writes dialplan. |
14:44.13 | Samot | It lets you make calls from the GUI |
14:44.20 | Samot | Offers Click to Call for web pages |
14:44.45 | Samot | Perhaps what you are looking at is a poorly implemented GUI to manage things. |
14:45.54 | Samot | So what version of Asterisk is this system running? |
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15:22.30 | Sibert | Samot, 11.13.1 |
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15:41.41 | igcewieling | a new attack? 144.91.82.224 - - [16/Dec/2019:10:36:48 -0500] "GET /prov/yealink/y000000000044.cfg HTTP/1.1" 404 228 "-" "libwww-perl/5.833" |
15:43.48 | Corydon76 | Do Yealink phones give up their configuration (including a password) to anyone? |
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16:14.02 | jkroon | Corydon76, no, but i've also started seeing attacks on MAC address based provisioning, which generally does contain MACs. So far these seem to be targeted brute-force. Meaning the attacker would have had to action some system-specific research first. Just hoping that the RPS style systems have some decent anti-hacking mechanisms in place. |
16:17.22 | Corydon76 | jkroon: yeah, I don't know. It's one of the reasons why I keep provisioned systems off the Wild West of public addresses. |
16:18.41 | Corydon76 | If I were ever in a situation where the provisioning was not onsite, I'd be strongly motivated to do the query over a VPN and protect both the provisoned system, as well as the provisioning system from random queries. |
16:33.22 | igcewieling | I would not be a problem if failed2ban didn't fail to ban them. |
16:45.34 | joepublic | in addition to fail2ban i have a few servers that run a homemade service that looks over netstat, picks out sketchy entries, bans then with iptables, and shares their ip addresses with each other |
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17:03.18 | jkroon | Corydon76, we have multiple layers of security to avoid issues. |
17:04.08 | jkroon | igcewieling, the one that scares me is a redirect coming off of an RPS. anything else I believe we've got covered. but still, multiple layers of safety is in order when dealing with the wild-west-web. |
17:04.42 | TandyUK | yealink RPS allows you to set a user/password on the http server, as well as fully encrypt the config files, so even if someone did manage to hack it, theyre not getting anything useful |
17:05.14 | TandyUK | and for a device to be given the user/pass/url, i have to register the device on there first |
17:05.33 | igcewieling | RPS? |
17:05.48 | TandyUK | redirection and provisioning service iirc |
17:05.54 | igcewieling | ag provisioning |
17:06.45 | igcewieling | we do provisioning via FTP so http prov attacks are an annoyance only. |
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19:53.37 | igcewieling | Does Intel no longer make network reference cards? Looks to me they only do chipsets now. |
19:55.01 | igcewieling | reference network cards. Like they do with motherboard chipets. I can buy an intel branded board with Intel chipset XXXYYY or I can buy some other company motherboard which uses the that chipset. |
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20:29.35 | drmessano | igcewieling: https://www.intel.com/content/www/us/en/products/network-io/ethernet/gigabit-adapters.html |
20:31.20 | drmessano | Gig-E adapters are basically obsolete now, so most of them are current but legacy |
20:44.11 | joepublic | if 10gb adapters were not priced as if made out of solid gold, gigabit adapters would be obsolete. |
20:54.38 | drmessano | Eh wat? |
20:55.48 | drmessano | $100 for a generic 10Gbe adapter |
20:57.22 | drmessano | Most motherboards have 1GBe onboard, so 10GBe adapters are addons. Only reason cheap 1GBe adapters even exist on the market is addons or replacements |
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20:58.58 | drmessano | But priced like they are mode of solid gold? Never seen that |
20:59.55 | file | 10G would actually be useful for me in a way, but I still shrug |
21:01.57 | drmessano | You can just about implement a 10GBe fiber link for a couple hundred bucks |
21:02.30 | igcewieling | I need to match the hardware in a failover server. |
21:03.38 | drmessano | and? |
21:04.20 | igcewieling | otherwise I might consider something faster |
21:04.42 | drmessano | So why can't you find it? |
21:05.12 | igcewieling | I was searching on the chipset I needed and was getting only intel chip docs and non-intel cards. |
21:05.33 | igcewieling | or ones which claimed to be intel, but didn't look like they were. |
21:06.05 | igcewieling | If I'd searched for something more generic I suspect I'd have found the cards without the link you kindly provided. |
21:06.05 | drmessano | Ehhh.. "didn't look like they were"... If you mean bulk packaging, welcome to 2019 |
21:06.19 | drmessano | I haven't seen intel cards in retail packaging in years.. brown box or plastic |
21:06.44 | igcewieling | I mean like I could not find the word "intel" anywhere on the pcboard in the pics. |
21:07.26 | drmessano | *shrug* what chipset is it? |
21:11.51 | igcewieling | 82576 I found what I was looking for. |
21:14.50 | drmessano | Yeah that's a common one |
21:16.12 | drmessano | However, it's out of date, hence the trouble finding it |
21:16.20 | drmessano | I was using those a decade ago |
21:16.49 | igcewieling | *nod* Plenty of non-intel branded cards using the chipset though. |
21:17.14 | drmessano | Right, because Intel doesn't make them anymore, I guess |
21:17.38 | drmessano | So it's not that they got out of the game, you're looking for an old card |
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23:04.41 | joepublic | Am i correct in assuming that an old-ish machine with a core 2 duo and 4gb ram will do nicely for low pbx call volume (12-ish extensions, 3-ish simultaneous calls)? |
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23:10.38 | Samot | yes |
23:11.25 | joepublic | thank you. |
23:12.34 | joepublic | nother clueless question, do sip to sip pass through the server or connect peer to peer? |
23:12.52 | joepublic | the word "calls" was supposed to be in there; please mentally insert it |
23:24.13 | electronic_eel | joepublic: sip usually passes through the server |
23:24.27 | electronic_eel | but sip is just the control channel, not the actual voice data |
23:24.44 | electronic_eel | voice is usually rtp |
23:25.22 | electronic_eel | rtp can pass throught the server (direct media option off) or connect peer to peer (direct media option on= |
23:25.24 | electronic_eel | ) |
23:26.44 | *** join/#asterisk Heart_And_Chops (~Heart_And@S0106cc2de0099182.no.shawcable.net) |
23:26.46 | drmessano | direct rtp almost never works if the peer is behind a firewall |
23:26.52 | drmessano | So don't even think about it |
23:27.51 | electronic_eel | his 12-ish extensions could do direct rtp for internal calls if they are within the same local net |
23:28.21 | electronic_eel | but running it through the server doesn't really add a lot of load |
23:30.18 | drmessano | I've never seen a need to run direct rtp in-house. |
23:30.56 | drmessano | If I have ever seen it implemented, it's putting endpoints and proxies together so one proxy isn't handling all the audio |
23:31.50 | *** join/#asterisk hfb (~hfb@47.139.16.213) |
23:33.11 | electronic_eel | I haven't had any problems with setting direct_media=off and letting asterisk handle all the audio |
23:46.10 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |