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06:08.50 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.30.0 (2019/12/23) 16.7.0 (2019/12/23) Standard: 17.1.0 (2019/12/23); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
06:22.30 | kakama | yeah naturally we've got passwords and network security measures, I just don't like getting hammered |
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06:48.03 | jkroon | kakama, my question to my sysads are always: are you willing to deploy this without a firewall, and without any retrofitted (fail2ban) security measures? If, and only if, the answer is a definitive yes, only then will I deploy. With a firewall, and with our equivalent of fail2ban. |
06:50.11 | jkroon | the only thing fail2ban really does is help protect against brute force attacks, and help saves you the resources associated with those, but your setup really shouldn't rely on just that. |
06:57.53 | kakama | secure password on our sip endpoint, don't think there's any surfaces to secure outside of that |
06:58.26 | kakama | usual server security and network segmentation ofc |
07:09.28 | tuxd00d | kakama: Will the phones be limited to static IPs or will the phones be on frequently changing IPs? |
07:09.52 | tuxd00d | Or will they all be in-house? |
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07:39.04 | Sekyourbox | Hi, When I follow example instructions, "configuring chan_pjsip". I try to connect with microSIP and get faild due to password |
07:39.19 | Sekyourbox | is there some type of char encoding that may be happening |
07:39.47 | Sekyourbox | tried manually typing and copy/paste. with and without trailing space |
07:40.23 | Sekyourbox | hits hitting the box correctly, but not sure what it should look like. Looks like the same as in the screenshot in the "registering phones" section of the manual |
07:43.09 | Sekyourbox | will sip clients work with pjsip? |
07:46.33 | Sekyourbox | sip accepts autodomain |
07:48.25 | Sekyourbox | says in the conf the default is username,ip.. Which is coming over as username@ip |
07:51.13 | kakama | oh we're linked to a SIP provider and a network of IAX2 users tuxd00d so we just forward those ports to the box, and a bunch of UDP ports for the voice data |
07:52.20 | tuxd00d | Then it would be best to block all outside SIP to only your provider's IPs. |
07:53.43 | tuxd00d | Sekyourbox: Client on same network as server? |
07:54.04 | Sekyourbox | yes, same broadcast domain |
07:54.16 | Sekyourbox | collision dom |
07:54.20 | Sekyourbox | what not |
07:54.30 | Sekyourbox | I can see it hitting the server |
07:54.36 | Sekyourbox | hence the password failure log |
07:54.55 | tuxd00d | That error can happen when SIP ALG is active on your router. |
07:55.04 | Sekyourbox | running cli + the latest version of ast that comes with astlinux |
07:55.34 | Sekyourbox | tuxd00d: not there yet, just testing the example in the manual |
07:56.52 | tuxd00d | https://github.com/irontec/sngrep <- Good friend to have |
07:57.09 | Sekyourbox | changed auth type to just read user name, same error in cli |
07:58.49 | tuxd00d | I'm not familiar with the manual you are referencing, so I'm not sure what you're seeing. |
07:59.02 | Sekyourbox | the official astrisk man |
07:59.15 | Sekyourbox | "creating sip accounts" |
07:59.31 | Sekyourbox | deployment>basic PBX Fun |
07:59.49 | Sekyourbox | failing to do "the most basic pbx" |
07:59.58 | Sekyourbox | shouldn't need pcap for that lol |
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08:00.28 | Sekyourbox | but my softphone is microsip |
08:00.32 | Sekyourbox | running on a 10 box |
08:00.56 | Sekyourbox | I'm new to this stuff, so not sure if pjsip should be working with regular SIP clients |
08:01.22 | tuxd00d | Well, there are many other open source or free SIP soft clients, wouldn't hurt to try another. |
08:01.23 | Sekyourbox | and the error is showing "chan_sip.c" |
08:01.50 | tuxd00d | It's just referencing the source file, where to find the error in the code. |
08:02.19 | Sekyourbox | does pjsip use chan_sip.c, or does regualr chan_sip.so use it? |
08:02.40 | Sekyourbox | I did an unload for the chan_sip.so just in case |
08:04.37 | Sekyourbox | but logs haven't changed their error when attempting to connect |
08:07.04 | tuxd00d | Does "pjsip show aors" show your accounts? |
08:08.30 | tuxd00d | "pjsip show auths" |
08:23.40 | Sekyourbox | shows both accounts as "unknown" |
08:23.50 | Sekyourbox | for the status |
08:23.53 | Sekyourbox | but it shows them |
08:24.29 | Sekyourbox | looking for a new clinet. mostly everything wants you to sign up for their "solution" |
08:24.39 | Sekyourbox | and I don't want a windows app |
08:25.28 | Sekyourbox | wait, not that command |
08:25.31 | Sekyourbox | lemme see |
08:26.58 | tuxd00d | Oh, your earlier questions about using a SIP client with PJSIP. PJSIP is just the name of the SIP library used. PJSIP and SIP are both SIP protocol; PJSIP just has more abilities. |
08:27.31 | Sekyourbox | yea 2 objects |
08:27.41 | Sekyourbox | k |
08:27.44 | Sekyourbox | wasn't sure |
08:27.59 | Sekyourbox | but there is a sip.so file and a different module |
08:28.09 | Sekyourbox | won't the defaults being loaded cause conflict? |
08:28.53 | tuxd00d | You shouldn't be loading the chan_sip module. |
08:29.14 | tuxd00d | And you need to make sure your pjsip.conf has transports configured |
08:29.27 | Sekyourbox | I copied and pasted from the example |
08:29.36 | Sekyourbox | I'll check for transports |
08:29.39 | Sekyourbox | what's that? |
08:30.23 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Transport+Selection |
08:31.07 | Sekyourbox | oh, the example has [transport-udp] |
08:32.36 | tuxd00d | Then your pjsip account you need "transport=transport-udp" |
08:33.30 | Sekyourbox | https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts |
08:33.38 | Sekyourbox | I literally copied and pasted this |
08:33.42 | Sekyourbox | example |
08:39.03 | tuxd00d | try "bind=0.0.0.0:5060" |
08:39.19 | tuxd00d | then "pjsip reload" |
08:40.20 | tuxd00d | Actually, you may have to reload asterisk as you're changing the transport. "core restart when convenient" |
08:41.36 | Sekyourbox | yea, I do that with my mods |
08:41.56 | Sekyourbox | would it drop auth due to a port? |
08:42.05 | Sekyourbox | not sure what that is trying to fix |
08:42.49 | tuxd00d | We want to ensure that chan_pjsip, not chan_sip, is answering on the port your using. |
08:43.33 | Sekyourbox | I did an unload for chan sip in modules.conf |
08:45.53 | Sekyourbox | no change |
08:46.03 | Sekyourbox | trying linephone |
08:53.16 | Sekyourbox | nothing showing up with linephone |
08:53.34 | tuxd00d | And you restarted asterisk? |
08:53.48 | tuxd00d | It sounds like Asterisk isn't listening |
08:54.36 | Sekyourbox | well its still showing in the logs with microsip |
08:54.52 | Sekyourbox | probably just how im setting it up. limited knowledge about what specifics I need |
08:55.03 | Sekyourbox | but following the manual seems I'm doing it right |
08:55.11 | Sekyourbox | gonna check the linephone manual |
08:55.16 | tuxd00d | I'm going to have to call it night. Ensure chan_pjsip is listing on the correct port, check logs, ... |
08:55.19 | tuxd00d | Good luck. |
08:55.23 | Sekyourbox | thx |
08:55.26 | Sekyourbox | ++ |
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13:40.55 | MICROburst | I'm stuck configuring some local lines, for example voicemail. https://paste.centos.org/view/46230021 I do not get any output from 'pjsip set logger on' |
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14:21.29 | Sekyourbox | ok, so I removed the ;comment from the manual, and now its workin |
14:21.38 | Sekyourbox | is that a bug in the manual? |
14:21.45 | Sekyourbox | manual bug? |
14:22.43 | Sekyourbox | I feel euphoria to have a green dot on my voip client though |
14:23.10 | Sekyourbox | "computers" |
14:35.34 | sibiria | if it's a bug that a commented directive is not in effect? |
14:38.29 | Sekyourbox | not being commented |
14:38.47 | Sekyourbox | or maybe its because I rebooted the server. user error more likely |
14:39.08 | Sekyourbox | I was reloading asterisk, not restarting |
14:39.23 | sibiria | some configuration changes may very well require that you restart asterisk |
14:39.31 | sibiria | rather than just the affected module |
14:39.42 | Sekyourbox | learning process |
14:39.55 | sibiria | so a simple 'core restart when convenient' will get the job done |
14:40.47 | Sekyourbox | Why is the example setting> aor_dynamic type=aor max_contacts=1 |
14:40.52 | Sekyourbox | when there are 2 contacts |
14:40.57 | Sekyourbox | shouldn't max contacts be 2? |
14:42.39 | Sekyourbox | https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts |
14:42.46 | Sekyourbox | in pjsip |
14:43.28 | sibiria | that setting relates to how many registrations each aor can hold |
14:44.01 | Sekyourbox | ok, because it was warning me in my vvvvvvvvvvvr cli |
14:44.03 | sibiria | it's not a limit for the number of contacts in total |
14:44.24 | Sekyourbox | as soon as the 2nd softphone was launched |
14:44.27 | sibiria | as an example, you may want to be connected from both the phone on your desk, and your mobile phone |
14:44.37 | sibiria | (to your account) |
14:44.50 | Sekyourbox | oh right, both softphones are on the same windows 10 test box |
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14:45.10 | Sekyourbox | is that the warning about the 1 limit? |
14:45.22 | sibiria | i don't know what the warning is :) |
14:45.26 | sibiria | but that's what the setting control |
14:46.28 | sibiria | another typical example would be a setup where multiple phones on different locations ring on an incoming call; first to pick up gets the call |
14:46.58 | Sekyourbox | "registration attempt from endpoint 'demo-alice' to AOR 'demo-alice' will exceed max contacts of 1 |
14:48.04 | sibiria | if you want to find better documentation of the options you can find them in pjsip.conf |
14:48.17 | sibiria | the default config *is* the full documentation, more or less |
14:48.46 | Sekyourbox | I think I read don't use it for the documentation |
14:49.11 | sibiria | it's certainly more complete than the wiki, but the wiki offers examples. i find that they complement eachother |
14:50.19 | Sekyourbox | cool cool |
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21:25.07 | joepublic | Well, the Hanlong Unicorn 3002 2-line analog ATA works a treat. |
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23:47.40 | wyoung | joepublic: That's a mouthful. |
23:48.12 | joepublic | I'm sure it sounds smoother in chinese |