| 00:02.03 | *** join/#asterisk K0HAX (~K0HAX@2604:5800:5:f005::20) |
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| 00:09.17 | *** join/#asterisk moe` (~moe@node-be028882.scarlet.cw) |
| 00:09.27 | moe` | hey guys |
| 00:09.47 | moe` | having problems with getting linphone to bria calls working... anyone know of any magic there? |
| 00:10.39 | moe` | the call establishes, but no audio |
| 00:11.09 | moe` | now before we say audio devices, etc, we know these work on both ends cuz we can call via skype connect out to landlines and it works for both bria and linphone |
| 00:11.20 | moe` | its literally linphone <-> bria is the problem |
| 00:13.21 | moe` | [TK]D-Fender are you around dude? |
| 00:14.11 | [TK]D-Fender | yup |
| 00:14.27 | moe` | any comments on the bria-linphone thing? |
| 00:14.29 | [TK]D-Fender | and the answer for this is generally always improper network setup |
| 00:14.41 | [TK]D-Fender | Stopping pinning names and start looking at DEBUG |
| 00:15.23 | moe` | ok but the same remote client, same box, running linphone ... no joy. running bria... joy. that's just a matter of how the client handles things then? |
| 00:15.43 | [TK]D-Fender | yes |
| 00:15.50 | [TK]D-Fender | One is making a WRONG assumption |
| 00:16.02 | [TK]D-Fender | Or compensating for your lack of properly configuring your server |
| 00:16.18 | moe` | ok, but when all remote clients are linphone it works |
| 00:16.25 | moe` | so bria is busted then, I gather |
| 00:16.28 | [TK]D-Fender | no |
| 00:16.45 | [TK]D-Fender | it may be COMPENSATING for your improper ASTERISK setup and covering your ass |
| 00:16.51 | [TK]D-Fender | Stopp guessing and start looking |
| 00:17.18 | [TK]D-Fender | No, bria is NOT "busted" |
| 00:17.50 | moe` | well I meant my bria config is busted |
| 00:18.10 | [TK]D-Fender | no. |
| 00:22.20 | moe` | I find it odd that even on each box and the asterisk server itself, if we drop firewall, bria to bria works, linphone to linphone works, but bria to linphone not. |
| 00:22.24 | *** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net) |
| 00:22.33 | moe` | looking at the asterisk logs, I see nothing that tells me why |
| 00:22.38 | moe` | but its not in debug mode |
| 00:23.14 | [TK]D-Fender | Stop saying "odd" and "broken". |
| 00:23.18 | [TK]D-Fender | You are jumping at excuses here |
| 00:23.47 | [TK]D-Fender | There is no "odd". There is nothing odd about this. |
| 00:23.55 | [TK]D-Fender | These clients are not magical. |
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| 00:33.49 | moe` | there is something stupid I am missing here |
| 00:34.24 | [TK]D-Fender | yes, 20 minutes later and still no debug |
| 00:34.26 | [TK]D-Fender | that's what's missing |
| 00:34.44 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-ktjgqwqqlubwqapd) |
| 00:39.16 | moe` | on "asterisk -rv" ... "core set debug 10" should be verbose eh? |
| 00:39.43 | [TK]D-Fender | useless |
| 00:39.46 | [TK]D-Fender | that is not SIP DEBUG |
| 00:39.54 | *** join/#asterisk eduzimrs (~eduzimrs@186.249.205.163) |
| 00:41.10 | moe` | gotcha |
| 00:41.55 | eduzimrs | Hi guys, im tryin to set up an * behind NAT with ISP failover, but i've set externip to ISP1, is it possible to turn this method automaticaly ? or every time i have to change the externip? i've been reading about externhost and NO-IP, is it the only way? |
| 00:42.26 | [TK]D-Fender | ediIf you're on a changing IP there's that and STUN. |
| 00:42.44 | WIMPy | Or script something. |
| 00:42.50 | [TK]D-Fender | yup |
| 00:43.43 | WIMPy | Some (longer) time ago I used to rewrite the externip of sip.conf in ip-up. |
| 00:44.28 | eduzimrs | but, using externhost to an FQDN and set externrefresh to the trick? |
| 00:44.46 | eduzimrs | but, using externhost to a FQDN and set externrefresh to the trick? |
| 00:44.51 | WIMPy | If that hostname is automatically updated... |
| 00:44.52 | eduzimrs | but, using externhost to a FQDN and set externrefresh do the trick? |
| 00:45.08 | eduzimrs | yah, i'd put and noip client |
| 00:45.22 | eduzimrs | yah, i'd put a noip client |
| 00:45.33 | eduzimrs | yah, i'd put a noip client running each 30s |
| 00:45.36 | WIMPy | Might still take some time. |
| 00:45.55 | eduzimrs | there is the time to update globaly |
| 00:45.56 | WIMPy | They will blacklist you. |
| 00:47.20 | eduzimrs | yah i havent think that |
| 00:48.07 | eduzimrs | externip does not support two ips right? |
| 00:48.21 | eduzimrs | that doesnt sound good by the way |
| 00:48.27 | WIMPy | right |
| 00:49.14 | eduzimrs | so, there is no way doing that using * options |
| 00:49.41 | WIMPy | Not with chan_sip. |
| 00:50.28 | eduzimrs | right |
| 00:52.13 | eduzimrs | maybe a script at cron using sed to substitute and sip reload |
| 00:52.18 | *** join/#asterisk bhans (~briguer20@221.253.69.146) |
| 00:59.07 | tompaw | Evening. |
| 01:00.56 | WIMPy | Or check if chan_pjsip can do better. |
| 01:36.27 | moe` | oh, got bria to one linphone client working |
| 01:36.39 | moe` | yeah [TK]D-Fender it is stupid settings, network |
| 01:36.48 | moe` | you da man |
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| 05:47.39 | moe` | I wonder how much fun it'll be to setup an inbound number via Skype for asterisk |
| 05:48.17 | moe` | got the outbound going, simply prefix 9 and out it goes |
| 05:48.44 | moe` | inbound could be fun, problem is skype offers numbers in areas I don't give a shit about |
| 06:06.42 | drmessano | Prefix 9? My god, why? |
| 06:13.07 | moe` | why not? |
| 06:13.22 | moe` | prefix 9 and whatever global number for outbound |
| 06:14.14 | moe` | hey, you're a yoda master of asterisk, is there a quick and dirty way to setup inbound and then once the call is connected the user must select an extension? |
| 06:18.24 | moe` | I can see multiple inbound numbers, but the plan is the same... the user must select an extension for a softphone. |
| 06:19.04 | moe` | that's it, that's all. no secondary ringing, no other crap, just call the number, select extension, period. |
| 06:19.47 | moe` | if I could get that fired up I'd have numbers in a few countries |
| 06:22.30 | moe` | drmessano, file, malcolmd, any suggestions for that? |
| 06:28.44 | moe` | should I actually need SRV DNS records? |
| 06:28.53 | moe` | I think not, no? |
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| 07:07.09 | Rico | hi |
| 07:08.21 | moe` | ok I have inbound working routed to a specific extension, any suggestions on how to have it go to a "menu" where a user has to input an extension? |
| 07:23.56 | ChannelZ | that's the whole points of the dialplan |
| 07:24.35 | ChannelZ | Make extensions in some context, then make the entry point for the calls Playback()/Background() a prompt telling them to enter something |
| 07:24.40 | ChannelZ | the rest is magic |
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| 07:59.30 | moe` | yeah the syntax for the conf files is what I'm after |
| 07:59.35 | moe` | I am impressed though |
| 07:59.51 | moe` | it was rather simple to direct an inbound number to an extension |
| 08:00.03 | moe` | asterisk++ |
| 08:00.36 | moe` | I dunno if any of the asterisk developers are here, but if they are, dudes, I owe you some beers. Damn good work. |
| 08:01.03 | moe` | the more I mess around with asterisk the more I am impressed |
| 08:02.23 | moe` | I started life in IT as a C programmer, so I at least I understand a little of what went into asterisk |
| 08:02.39 | moe` | thumbs up |
| 08:06.14 | moe` | ok now that I have an inbound, how do I setup so that users can call outbound appearing as from that same number? |
| 08:06.19 | moe` | asks the google machine |
| 08:06.47 | moe` | of course it has to do with my SIP uplink |
| 08:07.34 | wdoekes | moe`: your users have a context from where they start a call. somewhere from there, you'd set CALLERID(num)=xxx |
| 08:08.06 | moe` | and in what wonderful config would that be? |
| 08:08.34 | moe` | I'm working with one context only at the moment |
| 08:09.11 | wdoekes | extensions.conf obviously, where you define the dialplan |
| 08:10.49 | moe` | grep tells me its possible to define callerid in sip.conf as well, assuming I guess that the upstream SIP provider accepts it |
| 08:10.55 | moe` | ? |
| 08:12.22 | wdoekes | hah, the upstream sip provider, that's a matter of its own. there are lots of differences in how they want the callerid to be provided |
| 08:12.57 | moe` | in my case it's stupid skype connect |
| 08:12.59 | wdoekes | moe`: as for the callerid= option, yes: that defines the initial callerid when calling from that sip device |
| 08:13.36 | moe` | in that case, if the SIP upstream will accept it, I could set that globally |
| 08:13.40 | moe` | I assume |
| 08:13.41 | moe` | ? |
| 08:13.49 | wdoekes | I don't know about your particular ITSP, but you may need to tweak the sendrpid= option for that "device" |
| 08:14.46 | wdoekes | moe`: callerid will not be set globally, but you could use a template from which all your sip accounts inherit |
| 08:14.46 | moe` | right now, so far, I'm only plugged into skype |
| 08:15.09 | wdoekes | but more common is to do it from the dialplan, just before you Dial(SIP/your-skype-itsp/${EXTEN}) |
| 08:16.10 | wdoekes | or, if they take callerid in the From, you could cheat and set it as the fromuser= in the [your-skype-itsp] device config |
| 08:16.23 | moe` | ah, ok, so in my stupid case I have 9 prefix to use skype outbound, so using a different prefix I could adjust it to appear to come from different source(s) |
| 08:17.38 | wdoekes | I don't know what you mean by prefix |
| 08:18.26 | moe` | oh. when a client connects to my asterisk, they dial 9 and then the global number to get "out" |
| 08:18.56 | moe` | but my SIP link with skype, they're being cunts on the caller ID |
| 08:19.04 | moe` | must verify/register my company |
| 08:20.10 | moe` | my company is unixninja.net, you bastards :) |
| 08:20.14 | wdoekes | ah, like that. yes, you could set the CALLERID based on the that prefix. (e.g. ExecIf($["${EXTEN:0:1}"="9"]?Set(CALLERID(num)=123); Dial(SIP/your-itsp/${EXTEN:1}) ) |
| 08:20.42 | wdoekes | s/ the that / that / |
| 08:24.00 | stefan27 | Any new educated guess when 13.5 goes from rc to release? |
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| 08:28.33 | moe` | not understanding the expectations between 13.4 and 13.5, can I ask why? |
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| 09:02.53 | afournier | hi |
| 09:06.36 | afournier | i had problem in the past when loading asterisk "compiled later" modules. i.e building asterisk once without the module, installing it, cleaning the sources, rebuilding asterisk again with the module and then only installing the module, asterisk won't load it, until a complete install is redone... is it normal ? is it a known problem or security feature ? is it due to some compilation timestamp ? is there a way to bypass this ? |
| 09:09.33 | afournier | in fact, it's not really accurate, at both compilation time the module was built, but only installed the second time, menuselect.makeopts did not change from first compilation to the second |
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| 09:24.37 | afournier | it looks like /usr/sbin/asterisk and e |
| 09:25.14 | afournier | it looks like /usr/sbin/asterisk and /usr/lib/asterisk/<module_name_here>.so all contain the same md5 :) |
| 09:27.59 | afournier | ./build_tools/make_buildopts_h:BUILDSUM=`echo ${BUILDOPTS} | ${MD5} | cut -c1-32` |
| 09:28.28 | afournier | so it means BUILDOPTS changed during these compilations :/ |
| 09:29.22 | afournier | fair enough |
| 09:29.35 | wdoekes | :) |
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| 14:06.40 | Junior | hello! |
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| 14:07.47 | Junior | For linux (ubuntu 14.04) what piece of software (free) should i use for my sip accounts? (I need to be able to import contacts from a csv list and zoiper free doesn't do that). Thanks! |
| 14:08.33 | Junior | Meaning if anyone is willing to recommened something as i don't want to chose a wrong software. |
| 14:19.20 | mirela666 | Junior, I can recomend, Jitsi, Linphone, Twinkle |
| 14:28.32 | Junior | mirela666, thanks! I have downloaded jitsi and will continue with the rest |
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| 15:08.59 | hexanol | I have a question about CDR in Asterisk 13 |
| 15:09.04 | hexanol | In a scenario "Alice calls Bob then Bob answers", but where the call is setuped by an originate instead of Alice dialing Bob's extension |
| 15:09.12 | hexanol | The "billsec" in the resulting CDR includes the time Bob phone was ringing |
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| 15:09.18 | hexanol | Is this the expected behaviour ? Is this behaviour undefined ? |
| 15:26.51 | mjordan | billsec is from when the channel was answered. Period. It's based on the actual change in the channel state, so I'm not sure how that would be possible. |
| 15:27.05 | mjordan | If you think that is happening, I'd file an issue, but make sure you get logs |
| 15:27.10 | mjordan | including with 'cdr set debug on' |
| 15:27.22 | mjordan | but I'm pretty sure we're going to find out that the channel was answered when you didn't think it was |
| 15:30.18 | hexanol | the channel is answered indeed |
| 15:30.26 | mirela666 | hexanol, in the dialplan you are probably doing Answer() as within first step |
| 15:30.31 | mirela666 | steps* |
| 15:31.01 | mirela666 | that is setting the flag on channel than time for calculating billsec |
| 15:31.29 | mirela666 | hexanol, remove the Answer and leave the Dial application to set that flag on actual answer |
| 15:31.44 | hexanol | there's no explicit Answer in my dialplan |
| 15:31.50 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ohjdbjfuzpreaqbv) |
| 15:31.55 | hexanol | the behaviour comes from the fact I'm using an originate |
| 15:31.56 | mirela666 | hm |
| 15:32.16 | hexanol | i.e. if Alice calls Bob by dialing Bob's extension |
| 15:32.20 | mirela666 | then you might pastebin the dialplan |
| 15:32.38 | hexanol | than billsec is the time the conversation between Alice and Bob |
| 15:32.42 | hexanol | but if I do an originate |
| 15:32.56 | hexanol | i.e. "channel originate SIP/alice extension 1022@default" (1022 is Bob's extension) |
| 15:33.05 | mirela666 | oh |
| 15:33.12 | hexanol | then billsec is the time the conversation between Alice and Bob + the time Bob's ring |
| 15:33.24 | hexanol | because indeed, Alice's channel is answered... when she answers the originate |
| 15:34.15 | mirela666 | I'm not sure you can fix it with originate |
| 15:34.28 | mirela666 | maybe someone else has more exp with this |
| 15:35.11 | hexanol | what is a bit unfortunate is that in these case you can't really use the billsec |
| 15:35.45 | hexanol | to know how much time the conversation between Alice and Bob lasted |
| 15:38.37 | hexanol | that's why my question was "is this the expected behaviour" -- if it is, that's ok with me |
| 15:38.44 | hexanol | if it is not, then I could open a ticket on the tracker |
| 15:39.20 | mjordan | it is expected, as billsec is defined as being the time from when the Party A channel is answered to when the relationship between the Party A and Party B channel is dissolved for that CDR |
| 15:39.44 | hexanol | alright |
| 15:39.48 | hexanol | thank you for the explanation |
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| 15:55.34 | hexanol | there might be a small error in the "Blind transfer" example of the CDR specification |
| 15:55.52 | hexanol | in the second CDR, the duration and billsec values are not equal |
| 15:56.34 | hexanol | but Alice's channel was already answered the time SIP/charlie was called |
| 15:57.07 | hexanol | i.e. the start and answer value for the second CDR should also be equal (that's also what I have if I do the test) |
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| 17:49.10 | dwarf_lona | Hi guys, need some help on asterisk queue |
| 17:49.22 | *** join/#asterisk italorossi (~Adium@216.80.70.5) |
| 17:50.07 | dwarf_lona | SIP/5073 (dynamic) (On Hold) has taken no calls yet |
| 17:50.37 | dwarf_lona | what does On Hold means? |
| 17:54.22 | robmal | It means the member is paused. |
| 17:55.14 | dwarf_lona | Hi robmal, is it (paused) when the member is on pause? |
| 17:55.36 | dwarf_lona | my problem is i cant change the status of the membe |
| 17:55.38 | dwarf_lona | my problem is i cant change the status of the member |
| 17:55.52 | dwarf_lona | i tried to remove and add the member |
| 17:55.58 | robmal | queue unpause blahblahblah |
| 17:56.01 | dwarf_lona | i also tried to pause and unpause the member |
| 17:56.23 | dwarf_lona | i also tried to check the channel and hangup but no avail |
| 17:56.37 | dwarf_lona | any thoughts? |
| 18:04.34 | Milenco | what asterisk version are you using/ |
| 18:04.44 | Milenco | @ dwarf_lona |
| 18:05.03 | dwarf_lona | asterisk 10.9 |
| 18:06.06 | [TK]D-Fender | 10 = dead |
| 18:06.10 | [TK]D-Fender | upgrade immediately |
| 18:06.16 | [TK]D-Fender | 12 = dead. Don't go there either |
| 18:06.58 | pjensen00 | If you tell me that 13 is dead I'm doing to die too |
| 18:07.10 | Milenco | anyway dwarf_lona, from what i read the member reaches on hold state when it's in a call but that call is paused/on hold |
| 18:07.18 | Milenco | possibly because of a warm transfer |
| 18:07.30 | Milenco | either use asterisk 11 or asterisk 13 |
| 18:07.39 | dwarf_lona | is it an issue on 10.9? |
| 18:07.48 | Milenco | not sure, or even likely |
| 18:08.11 | Milenco | but its always hard to support outdated software versions, because many things got fixed in the newer ones.. |
| 18:08.17 | dwarf_lona | its just weird coz it doesnt happen before. |
| 18:08.34 | dwarf_lona | now it happen in random |
| 18:08.35 | Milenco | i'm still using asterisk 1.8 here and got weird bugs as well :( |
| 18:08.48 | Milenco | hard to believe its totally random |
| 18:08.59 | dwarf_lona | there are also times that after putting the call on hold, the calls hangup |
| 18:09.02 | Milenco | its possibly, but it usually should be reproducable |
| 18:09.08 | dwarf_lona | but that doesnt happen on all agents :( |
| 18:09.20 | dwarf_lona | yeah, me too |
| 18:09.31 | dwarf_lona | i tried to replicate the issue, but it doesnt happen to me |
| 18:09.50 | dwarf_lona | tried putting the call on hold for 2min, 5mins, and even 15mins |
| 18:09.52 | Milenco | can the problem be related to the phones themselve? |
| 18:09.57 | dwarf_lona | but I was able to go back to the call |
| 18:10.06 | dwarf_lona | i was thinking about that also |
| 18:10.07 | Milenco | so whats the issue exactly |
| 18:10.13 | dwarf_lona | they are using the phones Hold button |
| 18:10.44 | Milenco | you mean you put a new member in the queue and it enters with the on hold state? |
| 18:10.48 | Milenco | without an active call? |
| 18:11.00 | Milenco | (from the phones perspective) |
| 18:11.26 | dwarf_lona | first - when agent is engage on a call and put the cx on hold, when they unhold, call is already dead |
| 18:11.49 | dwarf_lona | second - when they join the queue they are already (on Hold) state |
| 18:11.59 | dwarf_lona | without any active call |
| 18:12.03 | *** join/#asterisk spicyramen_ (~Adium@173.227.7.2) |
| 18:12.13 | dwarf_lona | SIP/5073 (dynamic) (On Hold) has taken no calls yet |
| 18:12.23 | Milenco | did you verify there isnt a call still going on? |
| 18:12.36 | dwarf_lona | checked on active channel, and there's none |
| 18:12.53 | Milenco | you can manually hang up these stuck calls with 'channel request hangup' |
| 18:13.07 | dwarf_lona | that's actually what i do |
| 18:13.24 | dwarf_lona | but the thing with this particular problem is there no active channel to hangup |
| 18:13.40 | dwarf_lona | so im stuck with (onHold) state |
| 18:13.47 | Milenco | if asterisk doesnt see the channel and still shows the agent in 'on hold' state then its probably an asterisk bug |
| 18:13.59 | Milenco | although my knowledge is very limited... |
| 18:14.11 | dwarf_lona | understood |
| 18:14.17 | Milenco | this is a related issue from 2007 |
| 18:14.17 | Milenco | https://issues.asterisk.org/jira/browse/ASTERISK-10617 |
| 18:14.23 | dwarf_lona | how about on my first issue, any thoughts? |
| 18:14.30 | dwarf_lona | <PROTECTED> |
| 18:14.45 | Milenco | you should check with sip debugging on |
| 18:14.47 | dwarf_lona | but the channel is still active |
| 18:14.58 | Milenco | and see whats happening, although it can be a little hard to interpret |
| 18:15.16 | Milenco | easier might be to just check the verbose log for that call |
| 18:15.16 | dwarf_lona | yes haha |
| 18:15.33 | Milenco | inside asterisk config or in message logfile (if you have verbose logging to file specified) |
| 18:16.24 | Milenco | the first issue reeks for sip/network issues |
| 18:18.10 | dwarf_lona | my setup is - asterisk server is on a colocation. SIP users are on a remote location behind firewall |
| 18:18.31 | dwarf_lona | asterisk -- internet --- firewall --- users |
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| 21:15.22 | Cruz4prez | Is there a good recent comparision of AsteriskNow vs FreePBX distribution? All I'm finding is stuff thats several years old. |
| 21:16.06 | robmal | Everyone using GUIs died in the process. |
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| 21:53.09 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 22:09.46 | *** join/#asterisk spicyramen_ (~Adium@192.195.80.114) |
| 22:10.52 | spicyramen_ | hi people, Im using Asterisk 13.4.0 and getting Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. using SIPML5 |
| 22:10.58 | spicyramen_ | any ideas? |
| 22:13.01 | spicyramen_ | icesupport=true is true in rtp.conf |
| 22:28.23 | *** part/#asterisk mjordan (mjordan@nat/digium/x-bdevekcfsufjzasw) |
| 22:38.23 | spicyramen_ | stun show status showing (null) 0 30 3 INIT 0.0.0.0 0 |
| 22:48.15 | WIMPy | Cruz4prez: There's nothing to compare any more. |
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| 22:52.44 | Cruz4prez | WIMPy are you saying they are the same? Freepbx people say no with an overly vague answer saying they have no idea what asterisknow/digium does |
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| 22:53.39 | WIMPy | AsteriskNOW is now FreePBX only. The Asterisk GUI is dead as a Dodo. |
| 22:54.36 | Cruz4prez | Right, I know asterisknow uses freepbx, but what I'm asking are what other differences are there? what other packages? what other integrations? |
| 22:55.27 | WIMPy | I don't think that peole askin such questions are the target of either one. |
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| 22:56.51 | WIMPy | There was a rumour that the both distros are actually identical, but I don't know if that's true. |
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| 22:58.22 | Cruz4prez | yeah, many people seem to think that, but freepbx definitely seems to deny it. I know the freepbx iso is about 100 meg larger |
| 22:58.36 | robmal | Bitmaps. |
| 23:03.46 | spicyramen_ | my dear asterisk people, I just opened all udp and tcp in my asterisk 13.4 any idea |
| 23:04.06 | spicyramen_ | why stun show status shows: Hostname Port Period Retries Status ExternAddr ExternPort |
| 23:04.06 | spicyramen_ | (null) 0 30 3 INIT 0.0.0.0 0 |
| 23:05.30 | WIMPy | Free calls |
| 23:05.42 | WIMPy | No working STUN server? |
| 23:06.14 | spicyramen_ | <PROTECTED> |
| 23:06.14 | spicyramen_ | Module Description Use Count Status Support Level |
| 23:06.14 | spicyramen_ | res_stun_monitor.so STUN Network Monitor 0 Running core |
| 23:06.31 | spicyramen_ | thats correct, it shows: ip-172-31-51-6*CLI> stun show status |
| 23:06.31 | spicyramen_ | Hostname Port Period Retries Status ExternAddr ExternPort |
| 23:06.31 | spicyramen_ | (null) 0 30 3 INIT 0.0.0.0 0 |
| 23:07.40 | WIMPy | I've never used the stun module, so I don't know what it tell, but to me that looks like it was unable to contact any STUN server. |
| 23:10.18 | spicyramen_ | nvm I configured res_stun_monitor.conf |
| 23:10.21 | spicyramen_ | and now works |
| 23:10.25 | spicyramen_ | sorry user error |
| 23:10.32 | spicyramen_ | I though rtp.conf will be enough |
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