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| 05:29.07 | Akari | Hey so two questions |
| 05:29.50 | Akari | 1. is there a way to get dahdi to play nice when you're running a daemon not under root |
| 05:30.41 | Akari | 2. in the git build of Asterisk, is specification of the pin for MeetMe conferences not doable directly through the MeetMe() application anymore? I can do it through meetme.conf. |
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| 06:30.57 | drmessano | Akari: Meetme in pretty extinct |
| 06:31.01 | drmessano | Akari: Meetme is pretty extinct |
| 06:31.31 | drmessano | ConfBridge offers so much more |
| 06:31.47 | drmessano | and doesn't require DAHDI |
| 06:34.42 | Akari | I'm mainly just toying around with it for legacy purposes. |
| 06:36.19 | Akari | (Because it's in the O'Reilly book) |
| 06:37.19 | Akari | And in regards to question 1, I was concerned from the perspective of setting a system up with non-SIP trunks. |
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| 06:38.56 | Akari | But it would also be nice to know in the sense I wouldn't want to upgrade a system and inadvertently break something. |
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| 06:51.29 | Akari | The main problem with 1 is that the dahdi service seems to rewrite the permissions on the /dev/dahdi folder every time it's started. |
| 06:57.00 | *** join/#asterisk dpecka (~dpecka@fw4.i.cz) |
| 06:57.02 | dpecka | hello |
| 06:57.32 | dpecka | stupid Q: if I run asterisk with -F, where comes log from -vvvvvd ? |
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| 07:09.46 | bittis | goodmorning everyone |
| 07:10.32 | bittis | random non asterisk related question, been trying to locate a voip provider in china that i can communicate with in English, has anyone had any luck with such a quest? |
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| 07:13.20 | Akari | I think Voxbeam has an office in Hong Kong. |
| 07:13.31 | bittis | but no servers there |
| 07:13.39 | bittis | call quality is not so good |
| 07:15.34 | Akari | didlogic maybe? I've never used them. |
| 07:16.23 | bittis | let me look them up, thanks for the tip |
| 07:16.44 | bittis | tried a few local providers but it is difficult to communicate in English with them |
| 07:19.34 | Akari | I'd imagine. |
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| 08:44.10 | tzafrir | Asterisk fails to build on kfreebsd. It fails to run ldconfig for the libasteriskssl library. That we don't really need anyway. |
| 08:44.19 | tzafrir | oops, wrong window |
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| 09:10.56 | Guest0193 | Hello, what is the difference between asterisk & rasterisk? When grepping for asterisk in ps aux we get one asterisk and one rasterisk. Looks like there are two asterisks running on the same machine. |
| 09:13.02 | dpecka | Guest0193: man asterisk; man rasterisk |
| 09:13.17 | Guest0193 | Did not find any man entry for rasterisk |
| 09:13.45 | dpecka | so read a part about -x (? or -r) in man asterisk |
| 09:14.20 | Guest0193 | Oh, so if I run asterisk -rx 'sip reload' it will start an rasterisk process? |
| 09:15.04 | Guest0193 | The r flag should not do that |
| 09:15.13 | dpecka | -r Instead of running a new Asterisk process, attempt to connect to a running Asterisk process and provide a console interface for controlling it. |
| 09:16.56 | dpecka | use lsof or pfiles (depends on platform) to explicitely check, what each process has open, it will be obvious from ports |
| 09:18.08 | Akari | rasterisk is a symlink to asterisk on my installation. |
| 09:18.44 | Akari | But if you run it you'll notice it actually executes the same as asterisk -r |
| 09:19.05 | dpecka | yeah, it's like that everywhere, rasterisk (by its $0) triggers just asterisk -r |
| 09:20.33 | Akari | Sneaky |
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| 09:24.06 | dpecka | Akari: this is sneaky http://susepaste.org/view/raw/28732548 :D, not that |
| 09:25.17 | dpecka | ls -la |
| 09:27.45 | Akari | So are they all part of the same bin like busybox or something |
| 09:29.58 | Akari | Or are they hardlinked for some reason |
| 09:39.04 | dpecka | Akari: as you see hardlinked .... eg -inum for find |
| 09:39.15 | dpecka | they just operate based on the $0 |
| 09:40.23 | Akari | Ah, so a bit of both then. |
| 09:48.45 | Akari | Yeah that's pretty cool. I always forget the command name is an arg. |
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| 10:51.45 | danielyk | Hi, this OPTIONS Message does not look well. Is Asterisk able to do this better? http://pastebin.ca/3734756 |
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| 12:05.11 | akv | Hey, is it possible to get the name of a SIP account (incoming) from sip.conf as a variable in the dialplan? |
| 12:07.24 | leachim6 | hey folks |
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| 12:23.32 | Samot | akv: What do you mean "SIP account" |
| 12:26.00 | akv | Samot: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf - "tammari" (first line) is what I'm looking for |
| 12:26.30 | Samot | No. |
| 12:26.36 | akv | :( |
| 12:26.36 | Samot | That's the device peer name. |
| 12:27.12 | Samot | Why would you need to get your device names as a variable? |
| 12:28.12 | wdoekes | ${CHANNEL(peername)} or something, I guess? |
| 12:28.32 | wdoekes | check 'core show function ...' |
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| 12:29.45 | akv | Samot: currently we have a single incoming number, but I need to add an other - both should end up in the same queue and therefore the agents need to see which number is called along with the caller number |
| 12:30.15 | Samot | OK, that has nothing to do with the device peer name. |
| 12:30.49 | Samot | You need to match the new DID, just like you do the current DID and send it to the queue. |
| 12:31.45 | Samot | You can prefix the incoming Caller ID with something like <incoming DID>_ or <queue name>_ |
| 12:32.47 | akv | Samot: that's what i'm trying to do and it works, just need it to be dynamic, so it just works when a new number is added |
| 12:33.03 | Samot | What do you mean dynamic? |
| 12:33.37 | akv | add an other sip peer account in sip.conf and set the context to the same queue and woila, it works ;) |
| 12:34.02 | Samot | OK, you realize the the peer context is for how Asterisk treats calls from the device. |
| 12:34.27 | Samot | So if your Agents have the queue context that incoming callers use, it will treat calls from Agents like they are from the outside. |
| 12:35.27 | akv | ah, it's been way too many years since i've configged asterisk :/ The register lines in sip.conf is for the incoming calls |
| 12:36.50 | [TK]D-Fender | Register only tells the other side where to contact you and suggests the Contact:. It's up to the provider to choose to use that Contact: for calls sent, and you still need to process the incoming call against your peers, etc |
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| 13:19.52 | handel | do they will a asterisk box with sip built in that is easy, small, and for consumer use ? |
| 13:20.05 | handel | and one or two phone plugins? |
| 13:20.32 | handel | i want asterisk sip box |
| 13:20.42 | handel | cheap |
| 13:21.23 | Samot | There are no license or fees for Asterisk. |
| 13:21.34 | Samot | So you need to get a server and then it's your time to set it up. |
| 13:26.04 | [TK]D-Fender | "asterisk sip box" is not a thing |
| 13:26.21 | [TK]D-Fender | Please rephrase your request |
| 13:26.40 | [TK]D-Fender | <handel> and one or two phone plugins? <- also more specific on this |
| 13:26.48 | [TK]D-Fender | And he's out... |
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| 13:34.16 | Samot | Yup. |
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| 15:50.50 | overyander | do you need a separate aors for each endpoint? |
| 15:51.19 | overyander | or can an aor be shared like an transport or auth ? |
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| 16:02.30 | [TK]D-Fender | It makes no sense for 2 endpoints to have the same AOR |
| 16:02.42 | overyander | ok, thanks |
| 16:03.09 | overyander | i'm just reading the wiki and trying to learn pjsip |
| 16:03.09 | [TK]D-Fender | I account can be in many places, but it doesn't make sense for one uniquely identifiable contact to be 2 people. What would the channel be when they called? |
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| 16:52.15 | jrun | 'asterisk -nrx' still spits out ANSI escapes; any idea? |
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| 17:08.36 | overyander | what am i doing wrong here? i can't get a softphone to register through pjsip. here's the debug log http://pastebin.com/w3kUWSRQ |
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| 17:13.51 | igcewieling | jrun: how about asterisk -rx "blah" |
| 17:14.51 | [TK]D-Fender | [Oct 31 12:05:07] DEBUG[15520]: res_config_odbc.c:116 custom_prepare: Skip: 0; SQL: SELECT * FROM ps_auths WHERE id = ? |
| 17:15.10 | [TK]D-Fender | missing auths from the looks of it |
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| 17:15.26 | overyander | my auth is -- id: phone_auth, auth_type: userpass, password: mypass, username:my_voip_user. my aor: id: 1506, max_contacts: 3 my endpoints: id: 1506, aors: 1506, auth: phone_auth |
| 17:15.27 | [TK]D-Fender | You should probably be showing us the config to mattch... and standard pjsip debug |
| 17:15.30 | [TK]D-Fender | not "core |
| 17:15.32 | [TK]D-Fender | PASTEBIN it all |
| 17:18.09 | drab | hi, still trying to figure out this issue with asterisk loggin ending up in syslog. This is my logger show channels: http://paste.debian.net/891005/ |
| 17:18.28 | drab | as you can see syslog is not configured, but I still end up finding asterisk logs in /var/log/syslog |
| 17:20.17 | Akari | Do you have /etc/asterisk/logger.conf |
| 17:20.52 | drab | I do, and it's configured for the same exact stuff showing in logger show channels. Here it is: http://paste.debian.net/891002/ |
| 17:21.11 | drab | so logger configuration seems correct and not sending anything to syslog (the syslog0 line in logger.conf is commented out) |
| 17:23.06 | drab | the only guess I have is that asterisk is logging to stderr or something and systemd, which is starting it, is then taking those lines and sending them to syslog too |
| 17:23.17 | drab | but not sure how to check that and it's not configured like that in logger.conf |
| 17:24.43 | igcewieling | does it still show up in the syslog if you have the syslog daemon stopped? |
| 17:25.18 | drab | interesting question, I have not tried that, let me test |
| 17:25.33 | jrun | igcewieling: that's what i started with |
| 17:25.42 | Akari | You could also try commenting out the console part |
| 17:26.32 | Akari | That's the only one besides syslog.local0 that redirects the messages to a place other than a file. |
| 17:26.46 | jrun | igcewieling: got the escape chars and went back to find a way to disable it in man page. searching for 'colo' in man page i only see -W and -n and - |
| 17:26.50 | jrun | -B |
| 17:26.51 | Akari | You could also try explicitly specifying no options for syslog.local0. |
| 17:28.21 | jrun | i'm putting a script to create a fs tree of helps calling 'config show help ...' and when i get to the end of each *topic* i get those escape colours. |
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| 17:30.42 | jrun | Akari: that's for me? |
| 17:31.07 | Akari | No for drab |
| 17:32.11 | drab | ok the answer is yes, stuff ends up in syslog even if rsyslogd is stopped |
| 17:32.22 | drab | so stuff isn't being sent to syslog, it's being written to that file directly |
| 17:32.29 | drab | igcewieling: ^^ |
| 17:32.56 | drab | so it seems more and more like asterisk is outputing to stderr or something and systemd is logging that stuff directly to syslog |
| 17:33.55 | igcewieling | drab: I don't know of a fix, but at least you now know for SURE syslogd isn't involved. |
| 17:33.58 | jrun | this is the script which fails for odd chars like '*': |
| 17:34.05 | Akari | jrun: It depends on what kind of ansi escapes you're getting I suppose... Are you sure they're color escapes? |
| 17:34.06 | jrun | https://gist.github.com/f6e547254318c4b9f6e1f74cf45d1c27 |
| 17:34.17 | jrun | any suggestions is welcome. |
| 17:34.22 | drmessano | drab: How is asterisk being started? |
| 17:34.25 | jrun | Akari: yes |
| 17:35.13 | drab | drmessano: from systemd (standard way when intalling from packages) |
| 17:35.18 | jrun | obviously call the file something like conf_help.sh |
| 17:35.23 | drmessano | drab: and which file are we talkign about? |
| 17:35.28 | drab | and it's runnig as asterisk so it wouldn't even have permission to write to /var/log/syslog |
| 17:35.38 | igcewieling | I assume you mean "core show help" not "config show help" |
| 17:37.10 | drab | drmessano: /var/log/syslog is receiving the same content as /var/log/asterisk/messages |
| 17:37.17 | jrun | safer version so it doesn't nuke itself: |
| 17:37.21 | jrun | https://gist.github.com/0a08105301a84741b46df149829bf857 |
| 17:37.42 | drmessano | drab: is /var/log/asterisk/messages symlinked to /var/log/syslog? |
| 17:38.28 | drab | it is not: http://paste.debian.net/891157/ |
| 17:38.28 | jrun | it works nicely with vifm but that's about it :) would be nice to disable to escapes on demand. |
| 17:39.39 | Akari | jrun: The n flag only works if you're starting asterisk with it |
| 17:39.42 | Akari | Not if you're attaching later |
| 17:40.03 | Akari | So I'd say you should just sed the escapes out if you don't wanna. |
| 17:42.52 | drab | ok now I even tested with lsof and asterisk process most definitely does not have syslog open at all |
| 17:45.27 | Akari | See anything redirecting to logger perhaps? |
| 17:46.32 | drab | nope, no logger process, had checked for that |
| 17:50.53 | Akari | Hmm... Did you try commenting out the console bit of logger.conf? |
| 17:51.35 | drab | no, will try that next, have found something else to test with rsyslog that I wanna check first |
| 18:00.37 | drab | ok so my previous rsyslogd test was incorrect |
| 18:00.57 | drab | it was being socket activated even tho I thought I checked and didn't see a process respawning |
| 18:01.14 | drab | also when syslog down jornal still has the asterisk entry lines |
| 18:01.26 | igcewieling | syslog or rsyslog? |
| 18:01.47 | drab | so it seems pretty much certain at this point that asterisk process is putting on stdout/err the log lines and those are being captured by ssytemd, sent to the journal and from there to rsyslog |
| 18:01.56 | drab | rsyslog, writing to /var/log/syslog, that's what I meant |
| 18:02.34 | drab | while at the same time (the asterisk process) si also writing those same lines to /var/log/asterisk/full|messages as per configured in logger.conf |
| 18:03.34 | drab | so now the question is why does it do that and how to stop it since there doesn't seem to be a point in getting those logs twice |
| 18:03.43 | drab | also it makes the entire point of a syslog channel moot |
| 18:04.10 | drab | since it does that anyway, but maybe this is something on ubuntu/debian init scripts or something that you don't get if you compile from sources/don't use ssytemd, no clue |
| 18:04.29 | drmessano | Look at the actual init scripts |
| 18:04.39 | drmessano | No point specualting |
| 18:04.43 | drmessano | speculating |
| 18:07.18 | drab | yeah I was doing that |
| 18:07.34 | drab | and indeed asterisk is started with -f , -f Do not fork or detach from controlling terminal. |
| 18:07.52 | drab | so that would indeed put stdout/err on the terminal to be caught by systemd/the journal |
| 18:10.54 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 18:10.56 | Akari | You mean for the value of DAEMON? |
| 18:12.03 | igcewieling | that is entirely an distro issue. I've never seen that on CentOS, which is what I use. |
| 18:12.17 | drmessano | Indeed |
| 18:12.46 | drab | grep "^ExecStart" /lib/systemd/system/asterisk.service -> ExecStart=/usr/sbin/asterisk -g -f -U asterisk |
| 18:12.49 | drab | Akari: ^^^ |
| 18:12.57 | igcewieling | actually I see a few warnings in /var/log/messages, but nothing excessive. |
| 18:13.05 | drab | so yes, systemd service script for asterisk defines it |
| 18:13.14 | drab | see the -f in there |
| 18:13.36 | igcewieling | why are you obsessed with -f ? |
| 18:14.54 | drab | centralized logging, getting duplicates and having a hard time reviewing logs for asterisk |
| 18:15.36 | drab | I was setting up asterisk for syslog, that's how it started, and then started seeing all the dups and thought I broke something |
| 18:15.45 | drab | or there was something wrong in my conf |
| 18:17.25 | Akari | Hmm... I don't even have that file. |
| 18:18.12 | drab | Akari: are you on 16.04/xenial? |
| 18:18.24 | drab | if you are on 14.04 it still awsn't switched over to systemd iirc |
| 18:18.38 | drab | bbl |
| 18:18.53 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 18:19.06 | Akari | Yeah xenial. |
| 18:19.28 | Akari | I also don't see that file in the repo package. |
| 18:20.16 | drab | dpkg -S /lib/systemd/system/asterisk.service -> asterisk: /lib/systemd/system/asterisk.service |
| 18:20.23 | drab | ii asterisk 1:13.1.0~dfsg-1 amd64 Open Source Private Branch Exchange (PBX) |
| 18:20.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:20.57 | drab | it's a fresh install to test in a VM so have not made any customizations to it |
| 18:21.03 | drab | should be in the repo |
| 18:21.52 | drab | http://packages.ubuntu.com/xenial/amd64/asterisk/filelist |
| 18:21.56 | drab | it's ruight there |
| 18:22.00 | drab | 4th from the top |
| 18:22.22 | drab | anyway, really gotta go, thanks all for the help |
| 18:28.58 | Akari | Ah, found it |
| 18:30.18 | *** join/#asterisk cluelessperson (~cluelessp@unaffiliated/cluelessperson) |
| 18:30.34 | cluelessperson | Hi all, I'm using linphone, and there's an option for echo limiter, but it seems to be doing very LITTLE |
| 18:30.43 | cluelessperson | at around 1000, seemingly no effect |
| 18:30.48 | cluelessperson | around 200, seems to help |
| 18:30.52 | cluelessperson | 500? no effect |
| 18:30.56 | igcewieling | I recommend seeking support for linphone |
| 18:31.09 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 18:34.50 | igcewieling | if you think Asterisk errors are cryptic, Polycom is worse. "1031143116|sip |4|00|All the number of packets are used!! Packet tossed = 27486" |
| 18:38.02 | cluelessperson | christ |
| 18:41.57 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 19:02.55 | overyander | what am i doing wrong here? i can't get a softphone to register through pjsip. here's the debug log http://pastebin.com/pNPD1sQs. here's my config http://pastebin.com/ThbPndZr. I'm testing with Bria sofphone. it is configured as "User ID": 1506, Domain: 10.200.100.41, "Display Name": Voip_User and "Authorization Name": Voip_User |
| 19:07.05 | overyander | here's what i get from show commands for auths, aors, endpoints and transports. http://pastebin.com/fYJaRnNK |
| 19:12.01 | *** join/#asterisk monsterco (~monsterco@207.35.222.226) |
| 19:14.04 | *** join/#asterisk jennie (~chatzilla@unaffiliated/jennie) |
| 19:17.30 | jennie | hello, http://paste.ubuntu.com/23408178/ canyou please tell me whats going on here ? how to fix it? Asterisk shows connected in freepbx |
| 19:18.07 | Akari | overyander: You using SQL |
| 19:18.14 | overyander | jennie, are you using freepbx or core asterisk? |
| 19:18.17 | overyander | Akari yes |
| 19:18.44 | jennie | freepbx |
| 19:18.52 | overyander | running show commands shows the info from my tables, so i'm fairly certain it's connected and querying properly. |
| 19:19.10 | overyander | jennie you need to get help from the freepbx support |
| 19:19.21 | jennie | ok |
| 19:19.24 | Akari | overyander: What version of asterisk? 13.10 or maybe 13.11? |
| 19:19.36 | overyander | Akari 13.12.0 |
| 19:19.52 | Akari | Hmm |
| 19:20.24 | overyander | It's a fresh install. first time ever configuring pjsip, so i could have missed something simple and/or obvious |
| 19:21.11 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 19:22.29 | overyander | Akari, I'm correct in using "auth" instead of "outbound_auth" aren't I? |
| 19:45.13 | Akari | Yeah auth is right |
| 19:46.01 | *** join/#asterisk Akuma (~Akuma@aylmpq0104w-lp130-01-76-71-222-97.dsl.bell.ca) |
| 19:48.51 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:48.51 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:50.01 | overyander | is an "identify" section required or is that optional? |
| 19:50.19 | overyander | [Oct 31 14:49:04] DEBUG[20587]: res_pjsip_endpoint_identifier_ip.c:128 ip_identify: No identify sections to match against |
| 19:50.19 | overyander | [Oct 31 14:49:04] DEBUG[20587]: res_pjsip_endpoint_identifier_user.c:133 username_identify: Attempting identify by From username '1506' domain '10.200.100.41' |
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| 19:57.18 | overyander | [TK]D-Fender any ideas? |
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| 20:09.40 | file | overyander, um it's like Bria isn't seeing our challenge for authentication |
| 20:11.09 | *** join/#asterisk klow (~textual@96-81-150-137-static.hfc.comcastbusiness.net) |
| 20:13.49 | overyander | file, i see on line 300 http://pastebin.com/pNPD1sQs where * receives a reg request, then the debug looks up the info from the request and then replies on line 362 unauthorized. |
| 20:14.07 | file | yes |
| 20:14.13 | file | and we never get a new REGISTER with authentication |
| 20:14.34 | file | Bria appears to be retransmitting |
| 20:14.47 | overyander | would that be the msg on line 315 ? |
| 20:15.35 | file | 4, 79, 153, etc |
| 20:15.52 | file | the REGISTER is the same in each one |
| 20:16.04 | *** join/#asterisk matrix1233 (~matrix123@41.230.26.59) |
| 20:16.23 | overyander | is that related to [Oct 31 13:56:15] DEBUG[14851]: res_pjsip/pjsip_distributor.c:267 find_dialog: Could not find matching transaction for Request msg REGISTER/cseq=1 (rdata0x7f6e7c122728). ? |
| 20:16.42 | file | that's an internal debug message |
| 20:17.00 | file | that's just part of the handling process |
| 20:19.04 | overyander | ok |
| 20:19.51 | file | on the Asterisk side we are sending a response challenging for authentication, so Bria does not appear to be receiving it |
| 20:23.06 | overyander | file, using sngrep it doesn't appear that * is sending anything back to the server. http://pastebin.com/2LkzrvFJ |
| 20:23.53 | file | the message is displayed immediately before we put it on the socket to go out |
| 20:29.50 | overyander | file did you see that in my logs somewhere? even doing a tcpdump, i don't see any traffic leaving the asterisk box. http://pastebin.com/ULLq5M1N |
| 20:30.08 | file | I saw Asterisk responding in the log |
| 20:30.36 | file | line 362 |
| 20:30.49 | file | so either you've got a firewall problem on the machine itself, or something else is blocking it |
| 20:34.49 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 20:37.44 | Akari | Isn't that response a 401 tho |
| 20:38.02 | *** join/#asterisk adeel (~adeel@fw1.ridgeway.scc-zip.net) |
| 20:39.08 | Akari | To me the fact that the From and To addresses are identical seems suspicious |
| 20:39.34 | Akari | But I'm not sure. |
| 20:45.07 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 20:48.19 | Akari | Actually that part looks correct so yeah ignore me |
| 20:50.04 | overyander | lol |
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| 20:58.42 | overyander | I just noticed during a pjsip reload the following: The module 'res_pjsip_publish_asterisk.so' was not properly initialized. Before reloading the module, you must run "module load res_pjsip_publish_asterisk.so" and fix whatever is preventing the module from being initialized. |
| 20:59.11 | overyander | when i ran that command i got the following output: ERROR[5308]: res_pjsip_publish_asterisk.c:862 load_module: Entity ID is not set. |
| 20:59.31 | overyander | could that be related or just something else to fix later? |
| 21:01.43 | Akari | I think what file was saying was that the response from the Bria client was inappropriate |
| 21:02.12 | file | it's something to fix later |
| 21:02.22 | file | and there was no response from the Bria client, it never got our challenge |
| 21:02.43 | file | there's nothing in Asterisk that would prevent us from sending it when the message is output so it's outside of Asterisk |
| 21:03.16 | Akari | Just try pinging the client, for starters. |
| 21:09.21 | overyander | ping is fine |
| 21:12.39 | overyander | i get the same results using the bria app on my cell phone and routing over the public internet instead of local network |
| 21:13.24 | overyander | so, same results on pc using softphone on a separate vlan from the server as public internet using softphone on cellphone |
| 21:14.33 | overyander | both cases, i see in the pjsip log that it immediately replies back to the register request with a 401 unauthorized. both cases, i don't see any network traffic using either tcpdump filtering on port 5060 or using sngrep |
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| 21:15.54 | overyander | i have internet access, been doing yum updates, etc. i verified the ip, subnet, gateway, etc.. i have confirmed that iptables is off and firewalld is not installed |
| 21:16.31 | overyander | selinux is set to permissive |
| 21:24.28 | Akari | Yeah the response is rather fast coming back as well. |
| 21:24.40 | Akari | Arguably causal |
| 21:24.46 | Akari | It's possible you need a password. |
| 21:24.59 | Akari | Or that the server is expecting a password. |
| 21:25.13 | overyander | Akari where in my auth? |
| 21:25.32 | overyander | my auth has a password.. it's "pass" |
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| 22:50.44 | Akari | overyander: you should take a look at this if you haven't already. https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide#AsteriskPJSIPTroubleshootingGuide-AuthenticationNotAttempted |