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| 03:16.36 | life_of_e | Which DTMF mode triggers the choice to use the content type dtmf versus dtmf-relay? |
| 03:49.53 | [TK]D-Fender | life_of_e, Is this still regarding your HT Flash goal? |
| 04:34.28 | life_of_e | Yeah, I decided to poke around in the code |
| 04:34.57 | life_of_e | I think I figured out how to modify chan_sip but the details of how chan_pjsip generates the INFO packet eludes me right now |
| 04:37.30 | life_of_e | There's a branch point in chan_sip that changes the format of the packet depending on the type (dtmf vs dtmf-relay) |
| 04:38.31 | [TK]D-Fender | How many lines do you need to do this for and how much do you care? |
| 04:38.53 | life_of_e | Looks like it's one or two lines in chan_sip |
| 04:39.11 | life_of_e | chan_pjsip is harder to follow the code |
| 04:43.48 | life_of_e | I'm going to try it with chan_sip first just to see if it works. As far as I can tell from documentation I literally have to send '16' as the signal payload |
| 04:46.25 | life_of_e | In one branch that involved adding an extra else if to the if tree and in the other branch adding a new if else to choose between sending the requested digit 0-9A-F*# or sending a literal '16' |
| 04:51.03 | life_of_e | First step is figuring out how to run asterisk in the build directory without running make install so I can quickly test it. |
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| 11:28.53 | telekom_guy | Hi, is it possible to configure asterisk to dump output media data to a file? |
| 11:33.02 | file | that's not detailed enough to actually be able to answer, for example Record will record and store in a file, MixMonitor will record a call in progress |
| 11:34.36 | telekom_guy | Sorry, I'm new to asterisk. I need the Record functionality. Is it possible to do that by changing asterisk configuration? |
| 11:35.09 | file | yes, you configure the dialplan to do so |
| 11:35.34 | telekom_guy | thank you |
| 11:36.30 | telekom_guy | any links I can check out about that? |
| 11:37.28 | file | I don't have any really handy... |
| 11:37.44 | file | the thing about dialplan is that it's essentially a scripting language, so you need to know the basics and have context to understand how to use/modify it |
| 11:41.35 | telekom_guy | no problem, I'm a developer which is trying to use asterisk for incoming call (media type G711) and re-encode it to AMR-NB on outbound... I have AMR patched asterisk, all I'd like to do now is to record the AMR output... |
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| 12:20.20 | flok | any ideas why this doesn't work? https://pastebin.com/dn4s5DZs |
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| 12:21.41 | file | well.. all the dialplan is commented out for one thing, otherwise you'd have to state what "doesn't work" means |
| 12:23.36 | flok | file: yes, that's because I copied it in pastebin |
| 12:23.57 | flok | what doesn't work is that 172.29.0.11 cannot connect to 172.29.0.1 |
| 12:24.19 | flok | I see udp traffic, yet after 2.5 seconds 172.29.0.11 disconencts the calling partee |
| 12:24.37 | file | I'd suggest providing the console output on both sides including "iax2 set debug on" |
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| 12:30.40 | flok | so https://pastebin.com/7zJUY8Ph and https://pastebin.com/UkLXErKH |
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| 12:33.45 | Samot | == Everyone is busy/congested at this time (1:0/0/1) |
| 12:34.20 | flok | yet they aren't. the line is free. both sides can ping each other |
| 12:34.30 | file | it wants to authenticate you as 6100. |
| 12:34.40 | flok | that's strange, that's nowhere configured as such |
| 12:34.45 | *** join/#asterisk clearde (~clearde@217.163.90.130) |
| 12:34.49 | clearde | hello |
| 12:34.54 | clearde | im in need of urgent help |
| 12:35.17 | clearde | im running asterisk 13.7.2 on ubuntu 16.04 LTS |
| 12:35.24 | clearde | it was running perfectly for 2 years now |
| 12:35.35 | clearde | as of this morning unable to launch asterisk getting the following: |
| 12:35.53 | clearde | [ Initializing Custom Configuration Options ] |
| 12:35.53 | clearde | <PROTECTED> |
| 12:35.53 | clearde | Illegal instruction (core dumped) |
| 12:36.08 | Samot | And did you look at that file? |
| 12:36.16 | file | are you running in the cloud? |
| 12:36.19 | clearde | yes |
| 12:36.22 | clearde | running on Azure |
| 12:36.23 | *** join/#asterisk rpifan (~rpifan@p578D20EA.dip0.t-ipconnect.de) |
| 12:36.39 | file | you were migrated to a different host, and your Asterisk was compiled to run on the previous one |
| 12:36.47 | file | it is attempting to use instructions which aren't available |
| 12:36.57 | clearde | so how do i recompile to the new host? |
| 12:37.07 | flok | file: I doublechecked it and 172.29.0.1 has a user 6100 but that's a totally unrelated number. |
| 12:37.07 | file | disabling NATIVE_BUILD and rebuilding should make it work in the future even if migrated |
| 12:37.38 | clearde | how would i disable NATIVE_BUILD? |
| 12:38.02 | file | flok: it's probably because you're using "friend" and its finding that as an entry to challenge for |
| 12:38.12 | file | clearde: make menuselect and disable it in Compiler Options |
| 12:38.59 | clearde | @file thanks, rebuilding now |
| 12:39.07 | flok | lemme check |
| 12:39.15 | clearde | hoping it would work |
| 12:47.26 | flok | file: should it be user? on both sides? |
| 12:48.16 | clearde | if i do make && make install && make config does it override all my previous configurations? |
| 12:48.26 | file | generally separate entries are nice... you might also try adding "username=<username>" to the peer/friend |
| 12:48.28 | clearde | (mainly my extensions.conf) |
| 12:48.52 | file | clearde: only "make samples" and "make basic-pbx" touches /etc/asterisk |
| 12:49.08 | clearde | thank you |
| 12:57.25 | clearde | @file it worked, thank you so much |
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| 14:50.24 | life_of_e | Ok, so there's two sets of PJSIP files that appear to deal with SIP INFO, one is pjsua_app_cli.c and pjsua_call.c. Which one gets used during a SendDTMF() application call? |
| 14:50.37 | file | neither. |
| 14:50.40 | file | those are pjsua |
| 14:51.04 | file | INFO sending is in channels/chan_pjsip.c |
| 14:51.18 | file | specifically transmit_info_dtmf function |
| 14:52.03 | file | pjsua is a higher level API used for writing softphones pretty much, Asterisk doesn't use it |
| 14:52.19 | life_of_e | Ah ok |
| 14:52.38 | life_of_e | Strange, grep skipped that file when I did a search for "Signal=" |
| 14:52.52 | life_of_e | I see it now, thanks |
| 14:53.38 | life_of_e | So PJSIP always uses dtmf-relay? |
| 14:55.05 | life_of_e | I was trying to figure out how chan_sip chose dtmf vs. dtmf-relay but I don't see dtmf in chan_pjsip, only dtmf-relay |
| 14:56.25 | life_of_e | Is there a way to run Asterisk from the build directory for testing without installing? |
| 15:08.48 | seanbright | contrib/scripts/live_ast |
| 15:09.05 | seanbright | i don't know it's current maintenance status, so your mileage may vary |
| 15:09.09 | seanbright | its* |
| 15:10.41 | life_of_e | Thanks, I'll take a look at that |
| 15:11.27 | file | it only uses dtmf-relay |
| 15:17.42 | life_of_e | Thanks file. So that gets sent if dtmfmode=info for that endpoint otherwise it'll generate tones through the tone generator code, correct? |
| 15:18.18 | file | if rfc4733 then RTP out of band, if info then INFO, otherwise inband |
| 15:22.08 | life_of_e | Fantastic. I've already edited chan_sip so I'll experiment with that first this weekend and see what happens. If that works out I'll try modifying chan_pjsip and then if that works it'll be happy dance time. |
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| 17:04.01 | igcewieling | CDRs have a batch mode, does Asterisk have a batch mode for CELs? I can't find anything up to Asterisk 13, but maybe newer releases have the option? |
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| 17:07.28 | seanbright | negative |
| 17:07.45 | igcewieling | seanbright: thanks |
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| 18:07.10 | anexit | Can someone explain to me why I would be seeing 401 Unauthorized on INVITE requests? |
| 18:13.05 | [TK]D-Fender | Depends when |
| 18:13.23 | [TK]D-Fender | It's a common 1st response as a challenge for auth |
| 18:14.12 | anexit | Yeah, from what I'm seeing we're getting double 401's but on the third attempt the call works well |
| 18:14.21 | anexit | (for the user it's second attempt. |
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| 18:21.56 | seanbright | pastebin some logs |
| 18:23.12 | anexit | https://pastebin.com/WMcQR2rk |
| 18:23.18 | anexit | This is a digium system btw. |
| 18:23.52 | seanbright | then you should contact digium support, no? |
| 18:25.49 | seanbright | fwiw, that pastebin doesn't really have anything useful in it. you're saying you're seeing 401s but there is no sip logging there |
| 18:28.29 | anexit | yeah, the logging is terrible in digium (no offense) |
| 18:28.45 | anexit | The consultant believes it is the network |
| 18:28.50 | anexit | causing 401s |
| 18:29.07 | seanbright | well... there is no such thing as a "digium system" |
| 18:29.15 | seanbright | is it switchvox? |
| 18:29.20 | anexit | yes |
| 18:29.32 | seanbright | ok, so switchvox is mainly asterisk |
| 18:29.36 | seanbright | so 'sip set debug on' |
| 18:29.37 | anexit | indeed |
| 18:29.37 | seanbright | from the CLI |
| 18:29.45 | seanbright | that will turn on SIP logging |
| 18:29.55 | anexit | We don't have access to that box unfortantly |
| 18:30.04 | anexit | only switchvox is allowed |
| 18:30.05 | seanbright | well this has been fun |
| 18:30.09 | anexit | haha |
| 18:30.26 | seanbright | so what were you hoping to get out of this discussion exactly? |
| 18:30.36 | anexit | hope? |
| 18:30.43 | seanbright | we're all out of that |
| 18:30.54 | anexit | Appears I am too. |
| 18:31.17 | seanbright | typically you would see INVITE -> 401 -> INVITE with creds -> 200 OK |
| 18:31.42 | seanbright | without seeing the SIP log, we'd just be guessing |
| 18:32.00 | anexit | yeah this is REGISTER -> Invite -> 401 -> Register -> invite -> 401 |
| 18:32.10 | seanbright | that is not a thing |
| 18:32.13 | anexit | indeed, I'll make some calls |
| 18:32.43 | seanbright | if you are sending a REGISTER and getting back an INVITE, then we are truly in the end of days |
| 18:33.02 | anexit | maybe it's the other way around |
| 18:33.09 | anexit | ah yes |
| 18:33.11 | anexit | haha |
| 18:33.20 | seanbright | both of those are requests |
| 18:33.21 | anexit | There is no register |
| 18:33.25 | seanbright | ok |
| 18:33.33 | anexit | Invite -> 401 Invite -> 401 |
| 18:33.46 | anexit | then Invite trying, 200 |
| 18:33.57 | anexit | according to tcpdump |
| 18:34.29 | seanbright | if you can format the tcpdump such that it can be pastebin'd that'd be useful |
| 18:34.59 | anexit | how would you want it? |
| 18:35.42 | seanbright | so that the contents of the packets could be seen |
| 18:35.47 | seanbright | tcpdump -r capture.pcap -A |
| 18:36.05 | seanbright | (maybe... i'm have my own util to dump out pretty SIP captures) |
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| 18:43.51 | anexit | seanbright: what line is the password in sip? |
| 18:46.36 | seanbright | it's hashed with a nonce, so it shouldn't be in plaintext |
| 18:46.42 | seanbright | but it would be the Authorization header |
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| 19:38.47 | seanbright | good talk |
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| 23:46.40 | anexit | seanbright: no problem? |
| 23:47.09 | anexit | Sorry, got busy but I might have found the issue. Some reason asterisk does not like to fragment |
| 23:47.32 | anexit | DF is enabled which from what I read doesn't play nice with cisco firewalls |
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