| 14:16.34 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:16.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 16:40.04 | jrun | in res_pjsip; is there a way to disable sending maxptime in sdp message? |
| 16:42.00 | jrun | code.config seems to have max_ptime under [opus]; i'm wondering if that can be usen in [general] |
| 16:42.41 | jrun | but even there i don't see how it can be disabled. |
| 16:45.34 | jrun | max_packet_size is defined in res/res_pjsip_sdp_rtp.c; don't see any knobs for it :) |
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| 17:28.30 | jrun | any idea how to disable sending maxptime in pjsip? use_ptime=yes doesn't change anything either. |
| 17:28.38 | Samot | I despite Windstream. |
| 17:28.52 | Samot | s/despite/despise/ |
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| 17:40.25 | Samot | Although, this is one of the dandiest techs I've ever spoken to. |
| 17:40.54 | Samot | It's border line Ned Flanderish. |
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| 18:04.47 | igcewieling | OT, but can anyone recommend a decent domain registrar? Not eNom. |
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| 18:24.36 | Samot | Well I would say OpenSRS. |
| 18:24.43 | Samot | But they just bought eNom. |
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| 18:50.40 | danny_phonism | Hey all, I'm trying to get a list of extensions through AMI. FreePBX is generating longer password than AMI reports. The command I'm using is 'sip show users'. |
| 18:51.07 | danny_phonism | currently the secret reported is no longer than 15 characters through AMI. Is there an alternative command to get all extensions and the FULL secret? |
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| 19:12.34 | cervajs2 | @Digium https://issues.asterisk.org/jira/browse/ASTERISK-27037 there is link to SWP-9818 . can you share info if you have the same problem ? |
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| 19:15.51 | file | SWP is an internal issue for planning only |
| 19:15.56 | file | any comments and such go on the public issue |
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| 19:20.31 | Samot | cervajs2: Give them time... |
| 19:21.17 | file | I also have no timeframe on when it will get looked at by us (Digium) |
| 19:21.56 | Samot | You acknowledged it as a problem 6 hours ago. |
| 19:22.11 | Samot | I kinda figured that would be a good thing. |
| 19:22.27 | lorsungcu | file: if it's exploitable, does it get priority? |
| 19:22.43 | file | if it's a security issue in some way, yes |
| 19:22.58 | Samot | gets out his "how to hack asterisk" guide... |
| 19:24.22 | file | doesn't mean it would trigger a release immediately mind you, depends on things |
| 19:24.36 | Samot | I have faith. |
| 19:25.26 | lorsungcu | starts sending ugly registers to pjsip |
| 19:25.35 | Samot | And that wasn't sarcasm... |
| 19:28.15 | Samot | This is kind of a "Oh it hurts when I do X" issue and not really a "Hey, can someone take this metal shiv out of my aorta?" issue. |
| 19:28.38 | Samot | I think file's reaction would have been different. |
| 19:31.13 | cervajs2 | Samot: this is not issue which i discussed 6hr ago |
| 19:32.24 | file | it was an issue that was triaged a week ago |
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| 19:34.12 | cervajs2 | file: thank you. today, we will turn on pjsip again. on client side we upgraded jssip to latest version. we will see if core dump happens again |
| 19:34.49 | file | you can, you can add more info if you want - but that won't really impact when we would get to it (could be months away) |
| 19:35.05 | file | if you need it urgently then you'd need to see about getting a consultant |
| 19:35.44 | cervajs2 | file: i know/understand |
| 19:37.36 | cervajs2 | i'm trying find the root cause and scenario howto replicate. but tips/ideas from experienced guys can help. i'm not demanding your help |
| 19:38.09 | file | without digging into it there's nothing I can really offer |
| 19:40.04 | cervajs2 | ok no prob. i now understand what SWP-XXXX issues are for |
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| 20:20.19 | wabbits | https://news.fastcompany.com/even-telecom-workers-dont-want-to-talk-on-the-phone-4041400 |
| 20:20.59 | Samot | Not really news. |
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| 20:28.27 | salviadud | wabbits, they should just add a bunch of features that allow them to playback themselves |
| 20:28.37 | salviadud | Customers won't even know |
| 20:29.28 | wabbits | sip:echo@iptel.org you mean like this |
| 20:29.29 | wabbits | ? |
| 20:29.53 | salviadud | No, what I mean is this. |
| 20:30.04 | salviadud | Most of the job is saying the same stuff when somebody calls in. |
| 20:30.24 | salviadud | "Hello, this is John, welcome to... How can I help you?" |
| 20:30.41 | salviadud | well, John should record himself |
| 20:30.47 | salviadud | and with some help from features.conf |
| 20:30.57 | salviadud | John can hit *1 and blamo |
| 20:31.05 | wabbits | Well you have just revolutionized the industry. You may now retire to an island paradise. |
| 20:31.05 | salviadud | He does playback of his intro |
| 20:31.40 | salviadud | smiles |
| 20:31.42 | robmal | So... like this? http://money.cnn.com/2016/05/19/technology/watson-conference-call/index.html |
| 20:31.44 | wabbits | :) |
| 20:31.45 | salviadud | They should really do that |
| 20:33.07 | salviadud | robmal, yeah, something like that |
| 20:33.29 | wabbits | my boss used to have conference calls with 4 participants that all sat in the same cube quad. |
| 20:33.53 | salviadud | We could call it semi-automatic conferencing |
| 20:34.12 | salviadud | If you're spitting out the same stuff over and over again, just record it, and play it back |
| 20:34.37 | wabbits | reminds me of the joke about numbering the jokes in jail |
| 20:34.37 | salviadud | If you really need to jump in, you unmute or start talking |
| 20:35.18 | salviadud | I'd just do it like this |
| 20:35.32 | salviadud | confbridge with a computer running whichever iax or siphone you like |
| 20:35.40 | salviadud | and a flash soundboard |
| 20:35.52 | robmal | That's a great idea. But speech recognition is hard and IVRs are simple. So just make a recording for the problem and the solution for everything you know. |
| 20:35.57 | robmal | And you're fired :-) |
| 20:36.10 | salviadud | You still have to clic |
| 20:36.32 | drmessano | Comcast has already been doing that for years |
| 20:36.32 | robmal | If the computer can solve the issue by using playback - you're not needed anymore. |
| 20:36.35 | drmessano | Is your modem on fire? |
| 20:36.36 | salviadud | The advantage is not too much though, you still need a concious human behind the clics |
| 20:36.38 | drmessano | Press 1 |
| 20:36.44 | drmessano | "Ok call 911, then pee on it" |
| 20:37.16 | salviadud | The problem is the customer, and sometimes ppl want some sort of human behind the phone |
| 20:37.22 | robmal | salviadud: I assume you work 1st line support ;-) |
| 20:37.25 | drmessano | Oh thats easy |
| 20:37.31 | drmessano | Dont give them a human, ever |
| 20:37.34 | drmessano | Literally, fuck em |
| 20:37.35 | salviadud | O yes, I was young back then |
| 20:37.52 | drmessano | If they press 0, they go back to "Go" |
| 20:37.54 | salviadud | I forgot an h there. |
| 20:38.12 | salviadud | I was an international operator |
| 20:38.13 | drmessano | I've experience a few IVR's where there was ZERO, NADA, ZILCH human option |
| 20:38.15 | drmessano | Even with errors |
| 20:38.17 | robmal | You're doing it wrong. |
| 20:38.27 | salviadud | Dumbest job ever |
| 20:38.35 | salviadud | Had so much fun |
| 20:38.40 | robmal | Press 9 for next options. |
| 20:38.51 | robmal | Press # to go back |
| 20:39.00 | robmal | Goes back to the first 8 options. |
| 20:39.06 | robmal | ??? |
| 20:39.08 | robmal | PROFIT |
| 20:39.15 | salviadud | If the company is not worried about profit, then the IVR can suck. |
| 20:39.21 | drmessano | Yeah |
| 20:39.56 | salviadud | So, most goverment ivrs should be crappy. |
| 20:40.26 | drmessano | I cant remember which company it was.. But even the "Hit '0' 100 times" and whatever else didn't even work its way to an Indian |
| 20:40.36 | drmessano | First time i've run across that |
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| 20:42.25 | drmessano | My pride was hurt.. for years people called me "Tough Telephone Guy" because I could get anyone on the phone at any time to get anything resolved.. especially utility related issues overnight |
| 20:42.39 | drmessano | and TTG couldn't get a human :( |
| 20:46.18 | salviadud | Well, you can't win them all. |
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| 20:48.56 | wabbits | "Tough Telephone Guy" :) |
| 20:55.39 | drmessano | Yep |
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| 20:57.36 | drmessano | Got that nickname when I was still in high school.. Did "apprentice" (AKA Free) engineering work for a group of radio stations. They have a "Hotline", which is changed as needed. Well.. Ma Bell changed the number, as requested, but put a forwarding message on it |
| 20:58.22 | drmessano | So I managed to get Ma Bell to get a supervisor out of bed at 2am, drive to the CO, and login to change whatever he had to change |
| 20:58.42 | drmessano | After I was told "It cant be done tonight, sir" |
| 21:00.08 | drmessano | If you ever want to turn an underloaded Taco into dinner for two at Taco Bell, PM me for details lol j/k |
| 21:07.17 | wabbits | I guess we will have to take your word for it. |
| 21:08.55 | drmessano | Yeah, guess so |
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| 21:09.31 | drmessano | Customer is never wrong. Just have to remind them of that sometimes |
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| 21:13.23 | wabbits | Well if you got someone from the phone company out of bed at 2:00am, that is commendable. |
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| 23:15.48 | sawgood | [Jun 21 15:58:35] NOTICE[17699][C-0000e823] chan_sip.c: Call from '' (IP-Address:59930) to extension 'Phone-Number-1' rejected because extension not found in context 'default'. |
| 23:27.36 | sawgood | without running: sip set debug on (Asterisk 11.x) (is there a way) to find the INVITE for a call? |
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| 23:34.08 | sawgood | Asterisk does not record sip messages unless you have sip debug turned on. |
| 23:34.35 | salviadud | Where are you getting at sawgood ? |
| 23:35.17 | sawgood | I was trying to find an INVITE to a call which came to an Asterisk SBC about 10 min ago (and) I did not have sip set debug on at the time |
| 23:35.33 | Samot | No, that's not logged. |
| 23:46.52 | sawgood | [Jun 21 16:44:14] WARNING[17699][C-0000e8f4]: chan_sip.c:16587 check_auth: username mismatch, have ,<new-username>, digest has <old-username> |
| 23:47.47 | Samot | Can you replicate? |
| 23:48.28 | sawgood | every call (outbound) fails: with that error (but it was working) before I changed the [context] and accountcode= in sip.conf |
| 23:48.49 | Samot | OK |
| 23:48.54 | sawgood | the context and accountcode are the new-username (but) before the context and accountcode had the old username |
| 23:48.54 | Samot | So enable the sip debug |
| 23:48.55 | Samot | make a call |
| 23:48.58 | Samot | show the whole thing. |
| 23:49.04 | Samot | asterisk -rvvvvvvvvvv |
| 23:49.06 | Samot | sip set debug on |
| 23:49.19 | Samot | Snippets don't show anything of real value 95% of the time |
| 23:49.30 | Samot | ~pb |
| 23:49.31 | infobot | methinks pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:49.40 | drmessano | Did you reload? Lol |
| 23:50.08 | sawgood | yeah a few times (ha) |
| 23:50.48 | sawgood | [Jun 21 16:44:14] WARNING[17699][C-0000e8f4]: chan_sip.c:16587 check_auth: username mismatch, have <new-username>, digest has <old-username> |
| 23:51.04 | sawgood | [Jun 21 16:44:14] NOTICE[17699][C-0000e8f4]: chan_sip.c:25809 handle_request_invite: Failed to authenticate device "3212" <sip:new-username@65.49.22.226>;tag=E093C298-5E6 |
| 23:51.08 | Samot | GD |
| 23:51.12 | Samot | Show the WHOLE CALL |
| 23:51.20 | Samot | Stop posting snippets. |
| 23:51.40 | sawgood | when you say "whole call" you mean with sip set debug on |
| 23:51.56 | Samot | 7:49:03 PM <Samot> make a call |
| 23:51.57 | Samot | 7:49:06 PM <Samot> show the whole thing. |
| 23:51.57 | Samot | 7:49:12 PM <Samot> asterisk -rvvvvvvvvvv |
| 23:51.57 | Samot | 7:49:14 PM <Samot> sip set debug on |
| 23:51.57 | Samot | 7:49:27 PM <Samot> Snippets don't show anything of real value 95% of the time |
| 23:52.58 | sawgood | you know: I sort of would like a way for sip set debug on (to) be the only thing which comes up on the screen (instead) of these details and all the other signalling which flys up at the same time |
| 23:53.23 | sawgood | mixing calls from various people together ... in one SSH window |
| 23:53.36 | Samot | sip set debug ip <ipaddress> |
| 23:53.46 | Samot | sip set debug peer <peername> |
| 23:55.29 | sawgood | yep I love those commands to (but) with those running (the) normal SIP traffic from everyone else (is still) coming up on the screen (mixing) the way the sreeen looks (sort of hard to explain) |
| 23:55.42 | Samot | No, it's not. |
| 23:55.46 | Samot | I understand how this works. |
| 23:55.55 | Samot | You need to be able to show a full call |
| 23:55.58 | Samot | With a full debug. |
| 23:56.09 | Samot | That's the only way to troubleshoot this. |
| 23:56.15 | sawgood | maybe: have sip set dubug (not) come up on the screen, but rather dump to a text file |
| 23:56.23 | Samot | No. |
| 23:56.24 | Samot | GD. |
| 23:56.28 | Samot | How busy is this box? |
| 23:56.39 | sawgood | I've done that with great success in the past (dumped to a text file) |
| 23:56.47 | sawgood | maybe 40 or so active calls right now |
| 23:56.53 | Samot | That's it? |
| 23:56.57 | sawgood | somewhat-busy ... |
| 23:57.01 | Samot | No. |
| 23:57.13 | Samot | Well I guess if this is for a single business, sure. |
| 23:58.53 | sawgood | Samot: you ever use SiP workbench (small program) which reads .PCAPS and puts the whole call in order (from) the very 1st invite (all) the way to the last 200 (OK) BYE message |
| 23:59.00 | sawgood | sweet litle program |
| 23:59.06 | Samot | No. |
| 23:59.28 | Samot | If I really need something that bad, I use Wireshark |
| 23:59.34 | *** join/#asterisk oh4 (sid187181@gateway/web/irccloud.com/x-yinbuqtnndcqzhey) |
| 23:59.38 | Samot | But I rarely have to troubleshoot via PCAP |
| 23:59.46 | Samot | I just read the console output. |