IRC log for #asterisk on 20200119

06:08.50*** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net)
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
06:22.30kakamayeah naturally we've got passwords and network security measures, I just don't like getting hammered
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06:48.03jkroonkakama, 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.
06:50.11jkroonthe 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.
06:57.53kakamasecure password on our sip endpoint, don't think there's any surfaces to secure outside of that
06:58.26kakamausual server security and network segmentation ofc
07:09.28tuxd00dkakama: Will the phones be limited to static IPs or will the phones be on frequently changing IPs?
07:09.52tuxd00dOr will they all be in-house?
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07:39.04SekyourboxHi, When I follow example instructions, "configuring chan_pjsip". I try to connect with microSIP and get faild due to password
07:39.19Sekyourboxis there some type of char encoding that may be happening
07:39.47Sekyourboxtried manually typing and copy/paste. with and without trailing space
07:40.23Sekyourboxhits 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
07:43.09Sekyourboxwill sip clients work with pjsip?
07:46.33Sekyourboxsip accepts autodomain
07:48.25Sekyourboxsays in the conf the default is username,ip.. Which is coming over as username@ip
07:51.13kakamaoh 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
07:52.20tuxd00dThen it would be best to block all outside SIP to only your provider's IPs.
07:53.43tuxd00dSekyourbox: Client on same network as server?
07:54.04Sekyourboxyes, same broadcast domain
07:54.16Sekyourboxcollision dom
07:54.20Sekyourboxwhat not
07:54.30SekyourboxI can see it hitting the server
07:54.36Sekyourboxhence the password failure log
07:54.55tuxd00dThat error can happen when SIP ALG is active on your router.
07:55.04Sekyourboxrunning cli + the latest version of ast that comes with astlinux
07:55.34Sekyourboxtuxd00d: not there yet, just testing the example in the manual
07:56.52tuxd00dhttps://github.com/irontec/sngrep  <- Good friend to have
07:57.09Sekyourboxchanged auth type to just read user name, same error in cli
07:58.49tuxd00dI'm not familiar with the manual you are referencing, so I'm not sure what you're seeing.
07:59.02Sekyourboxthe official astrisk man
07:59.15Sekyourbox"creating sip accounts"
07:59.31Sekyourboxdeployment>basic PBX Fun
07:59.49Sekyourboxfailing to do "the most basic pbx"
07:59.58Sekyourboxshouldn't need pcap for that lol
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08:00.28Sekyourboxbut my softphone is microsip
08:00.32Sekyourboxrunning on a 10 box
08:00.56SekyourboxI'm new to this stuff, so not sure if pjsip should be working with regular SIP clients
08:01.22tuxd00dWell, there are many other open source or free SIP soft clients, wouldn't hurt to try another.
08:01.23Sekyourboxand the error is showing "chan_sip.c"
08:01.50tuxd00dIt's just referencing the source file, where to find the error in the code.
08:02.19Sekyourboxdoes pjsip use chan_sip.c, or does regualr chan_sip.so use it?
08:02.40SekyourboxI did an unload for the chan_sip.so just in case
08:04.37Sekyourboxbut logs haven't changed their error when attempting to connect
08:07.04tuxd00dDoes "pjsip show aors" show your accounts?
08:08.30tuxd00d"pjsip show auths"
08:23.40Sekyourboxshows both accounts as "unknown"
08:23.50Sekyourboxfor the status
08:23.53Sekyourboxbut it shows them
08:24.29Sekyourboxlooking for a new clinet. mostly everything wants you to sign up for their "solution"
08:24.39Sekyourboxand I don't want a windows app
08:25.28Sekyourboxwait, not that command
08:25.31Sekyourboxlemme see
08:26.58tuxd00dOh, 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.
08:27.31Sekyourboxyea 2 objects
08:27.41Sekyourboxk
08:27.44Sekyourboxwasn't sure
08:27.59Sekyourboxbut there is a sip.so file and a different module
08:28.09Sekyourboxwon't the defaults being loaded cause conflict?
08:28.53tuxd00dYou shouldn't be loading the chan_sip module.
08:29.14tuxd00dAnd you need to make sure your pjsip.conf has transports configured
08:29.27SekyourboxI copied and pasted from the example
08:29.36SekyourboxI'll check for transports
08:29.39Sekyourboxwhat's that?
08:30.23tuxd00dhttps://wiki.asterisk.org/wiki/display/AST/PJSIP+Transport+Selection
08:31.07Sekyourboxoh, the example has [transport-udp]
08:32.36tuxd00dThen your pjsip account you need "transport=transport-udp"
08:33.30Sekyourboxhttps://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts
08:33.38SekyourboxI literally copied and pasted this
08:33.42Sekyourboxexample
08:39.03tuxd00dtry "bind=0.0.0.0:5060"
08:39.19tuxd00dthen "pjsip reload"
08:40.20tuxd00dActually, you may have to reload asterisk as you're changing the transport.  "core restart when convenient"
08:41.36Sekyourboxyea, I do that with my mods
08:41.56Sekyourboxwould it drop auth due to a port?
08:42.05Sekyourboxnot sure what that is trying to fix
08:42.49tuxd00dWe want to ensure that chan_pjsip, not chan_sip, is answering on the port your using.
08:43.33SekyourboxI did an unload for chan sip in modules.conf
08:45.53Sekyourboxno change
08:46.03Sekyourboxtrying linephone
08:53.16Sekyourboxnothing showing up with linephone
08:53.34tuxd00dAnd you restarted asterisk?
08:53.48tuxd00dIt sounds like Asterisk isn't listening
08:54.36Sekyourboxwell its still showing in the logs with microsip
08:54.52Sekyourboxprobably just how im setting it up. limited knowledge about what specifics I need
08:55.03Sekyourboxbut following the manual seems I'm doing it right
08:55.11Sekyourboxgonna check the linephone manual
08:55.16tuxd00dI'm going to have to call it night.  Ensure chan_pjsip is listing on the correct port, check logs, ...
08:55.19tuxd00dGood luck.
08:55.23Sekyourboxthx
08:55.26Sekyourbox++
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13:40.55MICROburstI'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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14:21.29Sekyourboxok, so I removed the ;comment from the manual, and now its workin
14:21.38Sekyourboxis that a bug in the manual?
14:21.45Sekyourboxmanual bug?
14:22.43SekyourboxI feel euphoria to have a green dot on my voip client though
14:23.10Sekyourbox"computers"
14:35.34sibiriaif it's a bug that a commented directive is not in effect?
14:38.29Sekyourboxnot being commented
14:38.47Sekyourboxor maybe its because I rebooted the server. user error more likely
14:39.08SekyourboxI was reloading asterisk, not restarting
14:39.23sibiriasome configuration changes may very well require that you restart asterisk
14:39.31sibiriarather than just the affected module
14:39.42Sekyourboxlearning process
14:39.55sibiriaso a simple 'core restart when convenient' will get the job done
14:40.47SekyourboxWhy is the example setting>  aor_dynamic type=aor max_contacts=1
14:40.52Sekyourboxwhen there are 2 contacts
14:40.57Sekyourboxshouldn't max contacts be 2?
14:42.39Sekyourboxhttps://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts
14:42.46Sekyourboxin pjsip
14:43.28sibiriathat setting relates to how many registrations each aor can hold
14:44.01Sekyourboxok, because it was warning me in my vvvvvvvvvvvr cli
14:44.03sibiriait's not a limit for the number of contacts in total
14:44.24Sekyourboxas soon as the 2nd softphone was launched
14:44.27sibiriaas an example, you may want to be connected from both the phone on your desk, and your mobile phone
14:44.37sibiria(to your account)
14:44.50Sekyourboxoh right, both softphones are on the same windows 10 test box
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14:45.10Sekyourboxis that the warning about the 1 limit?
14:45.22sibiriai don't know what the warning is :)
14:45.26sibiriabut that's what the setting control
14:46.28sibiriaanother typical example would be a setup where multiple phones on different locations ring on an incoming call; first to pick up gets the call
14:46.58Sekyourbox"registration attempt from endpoint 'demo-alice' to AOR 'demo-alice' will exceed max contacts of 1
14:48.04sibiriaif you want to find better documentation of the options you can find them in pjsip.conf
14:48.17sibiriathe default config *is* the full documentation, more or less
14:48.46SekyourboxI think I read don't use it for the documentation
14:49.11sibiriait's certainly more complete than the wiki, but the wiki offers examples. i find that they complement eachother
14:50.19Sekyourboxcool cool
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21:25.07joepublicWell, the Hanlong Unicorn 3002 2-line analog ATA works a treat.
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23:47.40wyoungjoepublic: That's a mouthful.
23:48.12joepublicI'm sure it sounds smoother in chinese

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