| 00:00.21 | nglpx1 | I have sended the screenshot in group escrow, they believed me and gave me another half hour God bless you |
| 00:00.23 | Samot | Because this is beyond ridiculous at this point. |
| 00:02.00 | Samot | This could have been solved 10 frigging hours ago if you would have just done what was asked of you. |
| 00:02.34 | Samot | [Apr 26 23:43:34] NOTICE[5978][C-00000006]: chan_sip.c:26224 handle_request_invite: Call from '201' (5.171.201.70:48077) to extension '+393515201769' rejected because extension not found in context 'from-internal'. |
| 00:02.55 | Samot | ^^ Because that is one of those "huge glaring issues that stand out" things I said we would notice 10 hours ago. |
| 00:05.00 | nglpx1 | the pastebin you gave me to add is already in extensions.conf |
| 00:06.42 | *** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew) |
| 00:07.06 | Samot | No, it's not. |
| 00:07.19 | simplydrew_ | When performing a âsip show peer [extension]â for chan_sip endpoints, why does Reg. Contact show sip:100@[server_ip]:5060 instead of the actual extension? |
| 00:07.21 | Samot | I added a + to the pattern match and removed the + from the Dial string. |
| 00:07.35 | Samot | It's basically the same but allowing a + to be sent. |
| 00:07.49 | Samot | Again, Bria is sending +39XXXXXX there's no match. |
| 00:07.56 | Samot | Asterisk is rejecting the call. |
| 00:08.12 | nglpx1 | Okok, i try to make the modify and I send you the pastebin modified |
| 00:08.25 | Samot | Just add those lines to it. |
| 00:08.37 | Samot | simplydrew_: Because that's the contact. |
| 00:08.45 | Samot | simplydrew_: The contact is the location of the device. |
| 00:08.55 | Samot | simplydrew_: It's how Asterisk knows where to send things. |
| 00:09.42 | simplydrew_ | Samot: But yet two different extensions have 100@ and their endpoint IP address. Thatâs normal? |
| 00:09.55 | Samot | What do you mean by extensions? |
| 00:10.04 | Samot | Chan_sip peers? |
| 00:10.13 | simplydrew_ | Samot: Correct - two different peers |
| 00:10.32 | Samot | So they show the registered contact as 100@[same_ip]? |
| 00:10.51 | simplydrew_ | 100@whatever_phone_ip |
| 00:11.09 | Samot | Is one of the peers 100? |
| 00:11.18 | simplydrew_ | So I guess as part of the string, the endpoint IP is the thing that matters. I figured it would incriement by peer - 101, 102, etc |
| 00:11.19 | nglpx1 | Samot look if is correct https://pastebin.com/5EgyyQVY |
| 00:11.43 | Samot | nglpx1: I said add it, not replace what was there. |
| 00:12.13 | nglpx1 | Oh add.. okok wait |
| 00:12.41 | Samot | 7:59:57 PM <Samot> Just add that to the from-internal. <-- how was that not clear? |
| 00:15.19 | nglpx1 | https://pastebin.com/twfPwn1Z |
| 00:15.23 | nglpx1 | Like this? |
| 00:16.29 | Samot | Yeah. |
| 00:21.11 | *** join/#asterisk nglpx1 (05abc946@5.171.201.70) |
| 00:21.16 | nglpx1 | Samot |
| 00:21.32 | nglpx1 | Temporarily Unavaible 480 |
| 00:25.11 | *** join/#asterisk nglpx1 (05abc946@5.171.201.70) |
| 00:25.15 | nglpx1 | Samot |
| 00:25.19 | nglpx1 | I lost connection |
| 00:25.35 | Samot | Now shit |
| 00:25.43 | Samot | Its getting old |
| 00:25.52 | Samot | Show the failed call |
| 00:26.08 | nglpx1 | In the log debug? |
| 00:26.29 | Samot | Like before |
| 00:26.31 | Samot | Yes |
| 00:26.41 | Samot | Show me a full call that fails |
| 00:28.22 | nglpx1 | https://pastebin.com/fkYgZQdq |
| 00:30.25 | Samot | That call went out |
| 00:30.41 | Samot | They sent back 100 Trying |
| 00:30.55 | Samot | And a 183 Session in Progress |
| 00:31.10 | Samot | Then a 480..which means the other side didnt answer |
| 00:31.56 | nglpx1 | but I have the phone next to me |
| 00:32.24 | nglpx1 | 183 session? |
| 00:32.28 | Samot | Well |
| 00:32.39 | nglpx1 | my phone donât have recived that call |
| 00:32.44 | Samot | I also see a lot of transmissions to your IP |
| 00:32.54 | Samot | You have a crappy connection |
| 00:33.18 | *** join/#asterisk AsteriskRoss (~AsteriskR@37.157.48.2) |
| 00:33.21 | Samot | Is that phone registered to a different service? |
| 00:33.37 | nglpx1 | i am using 4g data |
| 00:33.59 | Samot | So you have the phone that should also receive this call? |
| 00:34.24 | nglpx1 | the phone that is to receive the call has a subscription with a traditional provider and one yes |
| 00:34.48 | Samot | You mean it is a copper line? |
| 00:35.07 | nglpx1 | yes I have it here |
| 00:35.21 | Samot | Then you need to call your provider |
| 00:35.33 | nglpx1 | 4g lte |
| 00:35.44 | Samot | The call was sent to them..why they returned a 480 you will have to ask |
| 00:36.23 | nglpx1 | but why, if I make the call directly by connecting bria to the provider, it goes out quietly |
| 00:36.50 | Samot | And the other phone gets the call? |
| 00:36.57 | nglpx1 | Yes sure |
| 00:37.10 | Samot | You need to ask the provider. |
| 00:42.21 | nglpx1 | and strange I don't understand the meaning |
| 00:42.28 | nglpx1 | Is* |
| 00:42.54 | nglpx1 | Itâs* |
| 00:45.07 | nglpx1 | it is even stranger that in 16 changing the librerire (which makes it unstable) makes me make calls wherever I want without problems |
| 00:45.16 | nglpx1 | Samot |
| 00:46.18 | nglpx1 | are you here? |
| 00:53.16 | *** join/#asterisk nglpx1 (05abc946@5.171.201.70) |
| 00:53.27 | nglpx1 | Samot  ? |
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| 14:05.19 | igcewieling | Is the chupahora gone? |
| 14:08.08 | Samot | Not sure what that means. |
| 14:11.21 | Samot | So, yes? |
| 14:12.15 | seanbright | support vampire? |
| 14:16.39 | igcewieling | chupacabra = goar sucker chupahora = timesucker |
| 14:17.37 | sibiria | and lets not forget chupa chups, the famous spanish brand of lollipops |
| 14:17.46 | sibiria | i.e. suckysuckers |
| 14:19.29 | seanbright | the only suckers i see are the ones helping him |
| 14:19.37 | seanbright | zing |
| 14:25.05 | Samot | Well unless it's coming from me, I don't help. I belittle. |
| 14:31.05 | HannaM | .. so, did someone really lost 16k euro yesterday? |
| 14:33.52 | seanbright | true, true |
| 14:34.15 | seanbright | i think it's like 70/30 |
| 14:40.18 | Samot | That was the claim, 16K bitcoin lost. |
| 14:41.07 | HannaM | yep - smells like some oc (organized crime) minion, if usa, italy and bitcoin are mentioned in the whole "discussion" |
| 14:41.51 | Samot | I doubt that. |
| 14:42.07 | Samot | Even crime orgs have an IT dept. and they got a budget. |
| 14:42.24 | HannaM | yeah, the big competent ones .. |
| 14:42.34 | Samot | This is most likely yet another crypto currency fanbois. |
| 14:42.42 | HannaM | this might be just the little backyard scam running somewhere .. |
| 14:43.16 | Samot | Because I see this all the time in networking communities. No idea what they are doing but gotta do the mining nodes because crypto mining is $$$ and "easy" |
| 14:44.10 | zamba | in asterisk 16.x, how do i do playback over agi? |
| 14:44.13 | zamba | exec? |
| 14:45.17 | seanbright | there is a command line command to get a list of AGI commands |
| 14:45.22 | seanbright | agi show commands |
| 14:45.25 | seanbright | or something like that |
| 14:46.07 | zamba | what does the dead column mean/indicate? |
| 14:46.25 | zamba | exec is dead, so :) |
| 14:46.26 | seanbright | if it can be used on a dead (hungup) channel |
| 14:46.35 | zamba | aha |
| 14:46.41 | seanbright | exec? |
| 14:46.50 | seanbright | how about 'stream file' |
| 14:49.44 | sibiria | STREAM FILE is the right method |
| 14:50.01 | sibiria | rather than 'EXEC Playback/Background' |
| 14:50.45 | Samot | Or GET OPTION if you're looking to playback and get DTMF |
| 14:51.10 | sibiria | STREAM FILE does the same, but without a timeout option |
| 14:57.39 | zamba | another question.. is there a simple sip proxy that i can place between my soft phone and the actual asterisk server? the problem is that the asterisk server is running on an ip i'm not able to connect directly.. right now i've NAT-ed some traffic on a neighbouring server (which is in the same layer-2 network and that i'm able to reach directly).. |
| 14:58.11 | zamba | this is currently for testing purposes.. the asterisk server will be relocated once we have it in production |
| 15:25.23 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
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| 16:34.40 | igcewieling | zamba: no, but there is a nice complicated sip proxy. It is called Kamailio. 8-| |
| 16:35.00 | igcewieling | You are better off getting a public IP address. |
| 16:39.23 | zamba | yeah, i've heard about that one |
| 16:40.36 | seanbright | i mean... it's not super complicated |
| 16:40.51 | seanbright | and i think someone implemented a nice UI frontend |
| 16:42.04 | zamba | hm.. i'll see if it's worth it.. |
| 16:42.10 | seanbright | and a sip proxy is not enough if you want the media to traverse the NAT as well |
| 16:42.17 | seanbright | you would need an rtp proxy as well |
| 16:42.24 | zamba | or just NAT it, right? |
| 16:42.41 | zamba | btw.., how do i set the language for sound files through AGI? |
| 16:42.52 | seanbright | dunno |
| 16:43.04 | seanbright | set the language of the channel i guess |
| 16:44.54 | sibiria | you can omit it if you want, or specify it explicitly for the soundfile |
| 16:45.26 | sibiria | e.g. you can play back "en/hello", or "hello" |
| 16:45.46 | sibiria | if the latter, asterisk will also try source the file from the sounds/ root directory |
| 16:47.54 | sibiria | LANGUAGE is the channel variable, btw. |
| 16:48.36 | sibiria | should be accessible with CHANNEL(language) |
| 16:55.36 | zamba | sibiria: i just set it in the dialplan.. i'm constantly second-guessing myself wheater i should put stuff in the dialplan or in an AGI script |
| 16:58.59 | igcewieling | zamba: if you need complex logic or for loops or while loops or to handle complex data like databsse lookups, use an AGI. |
| 16:59.54 | igcewieling | My primary AGI connects to a database, does a bunch of logic and lookups, then sets dialplan variables. My dialplan handles the actual dialing, etc. |
| 17:00.49 | igcewieling | This is very similar to how FreePBX does things. I suspect that is where I got the idea from, but I don't recall. |
| 17:01.17 | igcewieling | zamba: "core show applications" and "core show functions" are your friends. |
| 17:02.03 | Samot | Well considering this is being done on a FreePBX box... |
| 17:02.20 | igcewieling | looks around. This doesn't look like #FreePBX. |
| 17:02.28 | Samot | Why in the world haven't you looked at the Appointment Reminder module? |
| 17:03.19 | Samot | It might work enough for this. |
| 17:05.27 | zamba | Samot: appointment reminders seems to work on a schedule.. my thingy works by demand |
| 17:05.46 | Samot | Thingy. |
| 17:05.49 | Samot | Awesome. |
| 17:06.15 | zamba | right now that's what it is.. it's a PoC just barely standing on its own :) |
| 17:06.21 | Samot | So how many numbers can be in this call list? |
| 17:07.00 | zamba | Samot: that can be dynamic.. but it will be in a database.. so people who's out of town and such, won't be called |
| 17:07.14 | zamba | but i guess maybe up to 30-40 |
| 17:08.05 | Samot | OK and you're plan is to pull one number at a time and call it? |
| 17:08.30 | Samot | s/you're/your/ |
| 17:08.31 | zamba | well.. i think i will pull all numbers and then iterate over them one at a time |
| 17:08.35 | Samot | OK. |
| 17:08.57 | Samot | So what is the hangup here? |
| 17:09.06 | zamba | until one of them returns a positive feedback |
| 17:09.09 | zamba | what do you mean? |
| 17:09.32 | Samot | Let me ask another way. |
| 17:09.35 | zamba | why i'm not done yet? :) |
| 17:09.38 | Samot | Why does all this need to be AGI based? |
| 17:09.58 | zamba | i believe i answered that yesterday.. dialplan logic gives me a headache |
| 17:10.59 | zamba | doing stuff like response = agi.wait_for_digit(5000) was a breeze |
| 17:11.28 | zamba | and then jumping based on that |
| 17:11.55 | Samot | OK. |
| 17:17.32 | zamba | Samot: i'm currently using manager to initiate the outbound calls |
| 17:17.51 | Samot | OK and they have to hit a context... |
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| 18:03.26 | zamba | yeah, i have that ready.. context, exten and priority |
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| 19:37.03 | zamba | hm.. i have done some experimenting.. it seems like calling stuff from a hangup handler leads to some strange behaviour.. |
| 19:37.32 | igcewieling | zamba: depends on what you are doing. |
| 19:37.40 | zamba | it seems like this is in fact not called *after* it's hung up, but rather that it leads the call that should've been hung up in some kind of limbo |
| 19:37.56 | igcewieling | huh? |
| 19:37.58 | zamba | igcewieling: i want to run a script after a call has been hung up |
| 19:38.08 | zamba | that needs to run on its own |
| 19:38.11 | igcewieling | zamba: I do that on every call. |
| 19:38.27 | zamba | not sure what i'm doing wrong, then |
| 19:39.04 | igcewieling | My hangup handler calls an AGI which writes the various database entries required to close out call accounting and fraud mitigation. |
| 19:39.33 | zamba | https://dpaste.org/ag3z |
| 19:39.38 | igcewieling | If you are trying to do something stupid like trying to play audio to callers or run an IVR, that would be just crazy. |
| 19:39.59 | zamba | the dialplan is not polished at all.. it's currently just a PoC |
| 19:40.03 | zamba | so i'm just testing the building blocks |
| 19:40.29 | zamba | the hunt_employees.py script will start an asterisk manager and will start issuing new outgoing calls |
| 19:40.32 | igcewieling | Is hunt_employees.py an ACTUAL AGI or is it a simple python scrpt |
| 19:40.53 | zamba | igcewieling: what do you mean by actual agi? import asterisk.agi ; agi = asterisk.agi.AGI() ? |
| 19:40.58 | igcewieling | An ACTUAL AGI reads the STDIN data Asterisk sends the script. |
| 19:41.20 | zamba | well.. i'm starting to think that i may not need the stdin data from asterisk.. |
| 19:41.28 | igcewieling | zamba: noobs sometimes try to run regular scripts in Asterisk using AGI() |
| 19:41.29 | zamba | so maybe use System instead? |
| 19:42.05 | igcewieling | zamba: Do you need to do anything which interacts with asterisk on stdin/ardout? |
| 19:42.20 | igcewieling | Get dialplan variable? |
| 19:42.40 | zamba | igcewieling: well.. i was thinking i want/need the caller id from the person calling into the temp_callin, but i guess i can just pass that as an argument or write that to a file |
| 19:42.53 | zamba | igcewieling: so i don't really *need* something that interacts with asterisk |
| 19:43.43 | igcewieling | *nod* try using System without any AGI stuff. Manager is sometimes lumped under "AGI" but it isn't really. |
| 19:44.22 | zamba | gotcha |
| 19:47.33 | Samot | Huh. |
| 19:47.39 | Samot | It's like someone that does this said that. |
| 19:48.26 | zamba | maybe i'm a bit confused by my phone not hanging up.. i'm calling in from my cell phone.. and that keeps the connection open.. even though i've issued Hangup() from asterisk |
| 19:48.29 | zamba | is this normal? |
| 19:49.14 | Samot | No. |
| 19:49.32 | zamba | that's why i'm wondering if the hangup handler is to blame |
| 19:50.04 | zamba | i'll try without the hangup handler |
| 19:50.41 | zamba | yeah, then it closes the connection.. and the call is hung up on the cell phone side as well |
| 19:51.08 | zamba | so this thing is keeping the call open seen from the caller's side: Set(CHANNEL(hangup_handler_push) |
| 19:51.40 | Samot | AGI is analogous to CGI in Apache. AGI provides an interface between the Asterisk dialplan and an external program that wants to manipulate a channel in the dialplan. In general, the interface is synchronous - actions taken on a channel from an AGI block and do not return until the action is completed. |
| 20:07.11 | zamba | yeah.. but does that explain the non-hangup? |
| 20:07.25 | zamba | i've realized that you're not a fan of the hangup handler :) |
| 20:09.08 | zamba | so i'm starting to think about using cron jobs and touching files instead |
| 20:10.54 | zamba | to trigger something completely outside of asterisk |
| 20:20.27 | seanbright | if your AGI doesn't exit, neither will the channel |
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| 20:21.57 | zamba | seanbright: even if it's an hangup handler? which should be called *after* the channel is hung up? |
| 20:22.22 | seanbright | oh great point! |
| 20:22.25 | seanbright | yes, even then |
| 20:23.10 | seanbright | just make your AGI exit, i don't get what is difficult |
| 20:28.34 | Samot | Well the AGI can't exit until it's complete, I'm guessing. |
| 20:28.37 | *** join/#asterisk Tittan (be405fd2@190.64.95.210) |
| 20:29.06 | Samot | Which means it will stay open until the list is exhausted or an accept is submitted. |
| 20:32.18 | Samot | As always, hangup handlers, like the h extension, need to execute quickly because they are in the hangup sequence path of the call leg. Specific channel driver protocols like ISDN and SIP may not be able to handle excessive delays completing the hangup sequence. |
| 20:33.52 | Samot | I guess one could postulate that running an AGI during the hangup sequence that is going to linear dial 30+ destinations with at least a 25-30 second ring time and at least 30 seconds of call time would be considered an "excessive delay". |
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| 21:05.18 | igcewieling | A hangup handler is to handle hangups, not trigger dialing campaigns. |
| 21:08.29 | seanbright | i mean... it can |
| 21:08.38 | sibiria | it's just fine to use it to issue signaling |
| 21:08.50 | seanbright | just don't block. |
| 21:09.29 | igcewieling | I suppose I should not say "trigger", I should say "manage and run" |
| 21:09.37 | sibiria | can fork |
| 21:09.50 | seanbright | we'll accept that |
| 21:11.05 | igcewieling | Anyone know how much of a hanguphandler induced delay would cause SIP signaling issues? |
| 21:12.43 | seanbright | 11usec |
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| 21:13.01 | seanbright | (i do not) |
| 21:13.03 | igcewieling | .ignore seanbright |
| 21:13.05 | igcewieling | oopts |
| 21:13.12 | seanbright | happens |
| 21:19.28 | sibiria | the problem is more that asterisk stops responding to SIP traffic on the channel when it's hung-up. technically any delay can be too long delay, even a few milliseconds |
| 21:19.49 | sibiria | i still haven't gotten around to file the bug report about this that was discussed here some weeks ago... i'm a bit weighed-down with work etc. |
| 21:22.52 | Samot | How is that a bug? |
| 21:23.15 | sibiria | it depends on the scenario. in the case i brought up for discussion the call disposition can end up misjudged |
| 21:23.34 | Samot | Misjudge how? I missed this convo. |
| 21:24.39 | sibiria | asterisk can end up ignoring the last response from the opposite end and present no q.850 reason despite one arriving a millisecond later |
| 21:25.26 | Samot | Why would ASterisk need a q.850 reason for a hangup |
| 21:25.38 | Samot | Asterisk would be generating the BYE |
| 21:26.56 | Samot | You mean to the other channel? |
| 21:28.50 | sibiria | i'll make sure you let you know when i've written up the bug report |
| 21:29.09 | sibiria | to let you know* |
| 21:29.23 | Samot | Yeah, I guess I'm not following it right now so more details would be nice. |
| 22:10.04 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 22:10.12 | Kobaz | Anyone do any work with iSAC? |
| 22:21.28 | *** join/#asterisk Typhon (~Typhon@dslb-088-067-128-053.088.067.pools.vodafone-ip.de) |
| 22:54.24 | Samot | I have not. |
| 23:34.25 | *** join/#asterisk nglpx1 (05ab2e89@5.171.46.137) |
| 23:34.45 | nglpx1 | https://pastebin.com/7nSpQGUs |
| 23:35.00 | nglpx1 | Who can tell my why I recive this error? |
| 23:36.22 | Samot | Because the other side sent it. |
| 23:36.38 | nglpx1 | I have 13.8 today it worked fine, I made some calls everything is fine. now i saw that error and so i uploaded yesterday's snapshot where it worked and it goes anyway. |
| 23:37.10 | nglpx1 | What do you mean? |
| 23:37.35 | *** join/#asterisk AsteriskRoss (~AsteriskR@37.157.48.2) |
| 23:37.59 | Samot | I mean the side you sent the request to, sent back a 500 Server Error |
| 23:38.35 | nglpx1 | then the provider? |
| 23:39.50 | Samot | Correct. |
| 23:46.04 | *** join/#asterisk sinaowolabi (~Sina@102.134.114.1) |