| 00:02.56 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 00:05.56 | *** join/#asterisk bmg505 (~leon@196-209-225-224.dynamic.isadsl.co.za) |
| 00:06.13 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 00:09.40 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 00:12.56 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 00:27.57 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 00:41.16 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 00:44.24 | *** join/#asterisk jkroon_ (~jkroon@165.16.204.170) |
| 00:50.33 | *** join/#asterisk petris (petris@gateway/web/irccloud.com/x-heqrbfidsxlejvkt) |
| 00:52.56 | *** join/#asterisk Oooohboy (~Oooohboy@cpe-67-11-10-120.satx.res.rr.com) |
| 00:58.39 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-zcjjepqpksbxakpt) |
| 01:12.40 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 01:28.01 | *** join/#asterisk doop (~doop@colostomy.club) |
| 01:34.38 | *** join/#asterisk bmg505 (~leon@196-209-225-223.dynamic.isadsl.co.za) |
| 01:45.38 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 01:54.05 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 02:05.30 | *** join/#asterisk bmg505 (~leon@196-209-225-223.dynamic.isadsl.co.za) |
| 02:17.53 | *** join/#asterisk drmessano (drmessano@pdpc/supporter/active/drmessano) |
| 02:21.39 | *** join/#asterisk paulgrmn__ (~paulgrmn@184.75.212.67) |
| 02:31.26 | *** join/#asterisk bmg505 (~leon@196-209-225-223.dynamic.isadsl.co.za) |
| 03:17.55 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 03:18.55 | *** join/#asterisk sibyakin (~sibyakin@83.234.69.145) |
| 03:22.02 | molo | hi, all. |
| 03:24.18 | molo | I'm trying to get inbound calls working and having an issue. I've got registration and outbound working, but inbound INVITE messages don't seem to be processed. |
| 03:24.37 | molo | INVITE messages are definitely making it to the device. |
| 03:24.58 | molo | The INVITES are large and are fragmented between two packets. |
| 03:26.08 | molo | "sip set debug on" doesn't show the INVITE messages |
| 03:26.13 | molo | any ideas about how to further debug? |
| 03:29.28 | *** join/#asterisk boris_t (~boris_t@109.248.217.2) |
| 03:58.17 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 04:20.15 | *** join/#asterisk dobson (~dobson@68.ip-149-56-14.net) |
| 04:22.28 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 04:57.16 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 05:56.40 | *** join/#asterisk Phil88 (9a41363a@gateway/web/freenode/ip.154.65.54.58) |
| 05:56.54 | Phil88 | Hi everyone! I'm cautiously optimistic to get some assistance |
| 05:57.16 | Phil88 | Is anyone aware of issues with asterisk 13.13-cert4 that doesn't do repacketization? |
| 05:58.24 | Phil88 | Call comes in from upstream provider's trunk - routes to third party PBX and although the SDP parameters shows ptime=20 and maxptime=150 the actual packets are relayed with a 60ms refresh time |
| 05:58.56 | Phil88 | Checked all known bugs for this release and can't find anyone else that had the same issue, before I upgrade or downgrade I want to check if there's a way to work around this? |
| 06:08.33 | *** join/#asterisk Rogach (uid6398@gateway/web/irccloud.com/x-qzwduplwvdbgwdhu) |
| 06:22.35 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 06:34.25 | *** join/#asterisk znoteer_1 (~znoteer_@45.72.255.87) |
| 06:59.25 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 07:01.31 | *** join/#asterisk sibyakin (~sibyakin@83.234.69.145) |
| 07:39.30 | *** join/#asterisk boris_t (~boris_t@109.248.217.2) |
| 07:43.36 | *** join/#asterisk karelk (~karel@31.10.157.57) |
| 07:49.05 | *** join/#asterisk DanB (~DanB@clt-195.192.201.246.ip-anschluss.net) |
| 08:05.09 | *** join/#asterisk miralin (~Thunderbi@194.8.128.64) |
| 08:09.27 | *** join/#asterisk sekil (~sekil@cable-89-216-220-44.dynamic.sbb.rs) |
| 09:13.24 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:47.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:27.52 | *** join/#asterisk kbch (~kbch@178-82-169-0.dynamic.hispeed.ch) |
| 10:29.02 | kbch | I'm using Record with the Record Dialplan Application. It records Audio as .wav and Video as .h264. Is there an option to play the video, the .h264 file, on a Ubuntu Computer? |
| 10:36.17 | *** join/#asterisk nighty- (~nighty@s229123.ppp.asahi-net.or.jp) |
| 10:43.34 | stefan27 | I think that might be troublesome... but you could write your own format_mp4 module to asterisk |
| 10:44.24 | kbch | I don't need to do the conversion in Asterisk itself. The only thing I want is to playback the .h264 file I get from Asterisk's Record Applicaton |
| 10:44.39 | stefan27 | not sure what kind of non-asterisk stuff that can read the .h264 file that asterisk 13 produces |
| 10:45.25 | stefan27 | ah so you want asterisk to playback the file as well? not export the file to a different system? |
| 10:47.49 | kbch | no, I want to only export it. |
| 10:48.16 | kbch | In my application I basically need a screenshot from the "calling" video feed. |
| 10:48.29 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net) |
| 10:48.54 | kbch | So my plan is basically to make a short recording, convert frame from that somehow to a picture and send that off to my backend |
| 10:52.01 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 11:29.08 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 11:59.08 | *** join/#asterisk unoccupied (unoccupied@libreboot/developer/leah) |
| 11:59.50 | unoccupied | does anyone know if there's a libre way to connect a traditional telephone via laptop (e.g. on modem slot) using asterisk, connected to a SIP provider? |
| 12:00.11 | unoccupied | libre as in, no IP phones running proprietary software. just a phone plugged into a laptop |
| 12:01.08 | unoccupied | i'm not sure if a modem could be used for this, talking to asterisk. i'm using just a normal laptop |
| 12:03.45 | Samot | No. |
| 12:04.43 | unoccupied | what libre solutions could be recommended? |
| 12:05.01 | unoccupied | I might ask the FSF sysadmins about this, since i know they run asterisk in their office |
| 12:27.34 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:28.29 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:29.51 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 12:49.10 | *** join/#asterisk MrMojit0 (~MrMojit0@194.171.91.248) |
| 13:06.57 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 13:30.29 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 13:47.26 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:58.22 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:58.22 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:05.30 | *** join/#asterisk sebastienthiry (~Thunderbi@91.180.5.253) |
| 14:08.19 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 14:11.42 | *** join/#asterisk kharwell (kharwell@nat/digium/x-spythktrhvtsqufc) |
| 14:11.42 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:12.46 | *** join/#asterisk iaronson (~luigi@pool-100-15-95-237.washdc.fios.verizon.net) |
| 14:34.53 | *** join/#asterisk tafa2 (~tafa2@t.ldn.dsrtnet.com) |
| 14:37.46 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-svmujsrcsgyffvbp) |
| 14:37.46 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:41.43 | *** join/#asterisk sibyakin (~sibyakin@83.234.69.145) |
| 14:48.16 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ptzlmnbmkhgesysc) |
| 14:48.16 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:49.54 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 14:59.00 | *** join/#asterisk tehgooch (~tehgooch@unaffiliated/tehgooch) |
| 15:14.30 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 15:35.14 | afournier | on Asterisk 11 is there a way to activate an audio loopback on a channel (like the application Echo) but in the background ? |
| 15:35.30 | afournier | it's like doing the opposite of Echo cancelation |
| 15:39.17 | *** join/#asterisk sebastienthiry (~Thunderbi@91.176.205.167) |
| 15:54.37 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 15:59.20 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 16:00.09 | [TK]D-Fender | afournier, originate a new local channel using Bridge + Echo. |
| 16:06.33 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 16:24.41 | [TK]D-Fender | unoccupied, You can't use a modem on a laptop for this. That port is to plug a LINE, not a PHONE. Nothing is going to make and "output" device into a proper "input" device |
| 16:26.34 | *** join/#asterisk danhardman (~danhardma@80.194.255.137) |
| 16:28.50 | danhardman | Hi all, I'm totally new to asterisk and phone systems in general. I was hoping someone could point me in the right direction to read up on what I want to achieve? |
| 16:30.49 | doop | ~book |
| 16:30.50 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:31.06 | doop | danhardman, take a look at that |
| 16:31.11 | *** join/#asterisk yteltom (~textual@208.74.138.242) |
| 16:31.14 | danhardman | I want to be able to receive calls on a linux server and receive/send audio to the call programmatically in realtime |
| 16:31.27 | doop | danhardman that's pretty basic and easy |
| 16:31.34 | danhardman | doop, I'm already on it! Quickly realised asterisk is very big haha |
| 16:32.44 | danhardman | Is there any terminology for what I'm trying to create? |
| 16:32.54 | [TK]D-Fender | "to the call" <- from where? |
| 16:34.05 | [TK]D-Fender | we're going to need more details on exactly what you're trying to accomplish |
| 16:34.40 | danhardman | So I have a UDP socket setup and it's sending and receiving audio. I want to be able to proxy audio from a phone line to and from that UDP stream |
| 16:34.57 | [TK]D-Fender | * is not a proxy |
| 16:35.02 | *** join/#asterisk sekil (~sekil@cable-89-216-220-44.dynamic.sbb.rs) |
| 16:35.08 | danhardman | So when someone calls in and starts speaking, I can send that voice down the UDP stream and visa versa |
| 16:35.15 | [TK]D-Fender | So if you're hoping for MITM that is not what * does |
| 16:36.05 | [TK]D-Fender | * is a B2BUA. |
| 16:36.26 | danhardman | I guess it would be MITM, but I'm not expecting to connect the call to another phone, I want my server to be the phone?? Sorry I really have no idea about phone systems |
| 16:36.41 | [TK]D-Fender | And can't just take raw UDP audio. A call setup is required unless you're writing a very custom module yourself |
| 16:36.59 | [TK]D-Fender | You need to have a better idea of the path you're trying to implement |
| 16:37.05 | [TK]D-Fender | And description of how the pieces talk |
| 16:37.15 | *** part/#asterisk igcewieling (~ewieling@speedy-02.nyigc.net) |
| 16:37.22 | [TK]D-Fender | If you don't ahve that we can't confirmif or how * might be able to accomplish it |
| 16:39.03 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 16:40.35 | *** join/#asterisk wearyhacker (~wearyhack@4.13.169.217.in-addr.arpa) |
| 16:44.16 | unoccupied | [TK]D-Fender: thanks. i'm unfamiliar so i wasn't sure, that's why i asked. thank you for confirming. |
| 16:44.37 | [TK]D-Fender | You're welcome |
| 16:44.42 | unoccupied | as stated in my original messages, I also said I'd ask the FSF sysadmins what they use, believing that they use asterisk. I received a reply, though not one that is helpful to me. |
| 16:44.56 | unoccupied | I'll shop around and see what I can find. thank you for your help, nonetheless. |
| 16:45.22 | unoccupied | (turns out FSF doesn't use asterisk. they use... normal phones. lol) |
| 16:45.36 | unoccupied | (landline) |
| 16:46.43 | unoccupied | my motivation is a totally libre phone system. currently, i use a mobile phone, but that's problematic: even with free OS like replicant, the bootloader etc is non-free as is the baseband processor which has access to main cpu/ram and is potentially updateable by the telco |
| 16:47.02 | *** join/#asterisk fblackburn (~fblackbur@modemcable094.94-70-69.static.videotron.ca) |
| 16:47.08 | unoccupied | i found IP phones and whatnot online, including things that can be used with asterisk, but the issue I always find is they have some sort of firmware on the handsets |
| 16:47.27 | [TK]D-Fender | There are a tiny handful of USB-FXS adapters out there if you insist on connecting an analog phone to your laptop. hard come-by and trick to get working with other software |
| 16:47.35 | unoccupied | what i *can* do is install linphone or GNU Ring in debian, and use that with a SPI provider. that's probably what i'll end up doing, with a cheap little mic headset |
| 16:48.02 | [TK]D-Fender | and yes, IP phones have to have firmware .. they are computer devices |
| 16:48.11 | unoccupied | [TK]D-Fender: USB might be acceptable, for that use-case scenario. though if i did that it'd be pointless |
| 16:48.29 | unoccupied | at that point i may aswell just use ring/linphone with a headset on the line in/out jacks of my laptop |
| 16:48.36 | [TK]D-Fender | What is your real objective and reason for your approach? |
| 16:48.51 | unoccupied | in terms of software/tech: everything free software |
| 16:49.12 | [TK]D-Fender | Well the software side can be free as far as laptop side is concerned. |
| 16:49.17 | unoccupied | in terms of everything else: mobile phones suck. i want to get rid of my mobile phone, and use my laptop (a libreboot laptop, running debian) as a phone, connected through a SIP provider |
| 16:49.31 | *** join/#asterisk salviadud (~ralfalfa@187-162-28-201.static.axtel.net) |
| 16:49.34 | [TK]D-Fender | I'd just use a softphone with a bluetooth headset then |
| 16:49.37 | unoccupied | i've done this before. i know how to do this... linphone/ring can be used with sip provider |
| 16:50.09 | unoccupied | this is the free-software way to do it, because then you just use a cheap headset and you're done. headsets are just dumb devices... a little speaker in my ear, and a mic to my mouth |
| 16:50.18 | unoccupied | and the software/hardware side is libre |
| 16:50.39 | unoccupied | i wanted a more traditional "pick up a telephone, dial it" sort of thing that is libre. but if that's not possible. i'll use the above method instead then |
| 16:51.03 | [TK]D-Fender | What I describes with USB FXS can work if you can find one. |
| 16:51.08 | [TK]D-Fender | It's a very tiny market. |
| 16:51.11 | unoccupied | besides that, the other motivation is cost. calls through SIP providers are cheap because they maintain less infrastructure |
| 16:51.19 | unoccupied | Yes, I'll look into USB FXS as you suggest |
| 16:51.31 | [TK]D-Fender | What is the driving reason for it being "libre"? |
| 16:51.43 | unoccupied | Because I'm basically Richard Stallman's lost daughter. that's why |
| 16:51.47 | unoccupied | i want all my shit to be free :P |
| 16:52.06 | unoccupied | ideological reasons mostly. but also security reasons. |
| 16:52.46 | unoccupied | mobile phones are pwned by default in terms of security. not trustworthy at all |
| 16:52.47 | [TK]D-Fender | Well you pay for hardware. the fact it may have firmware for ATA's for example and that it is "closed" source ... is this a serious philosophical issue for you? |
| 16:52.51 | unoccupied | landline services are expensive |
| 16:53.02 | unoccupied | [TK]D-Fender: yeah that's why i'm not getting an ATA either |
| 16:53.56 | unoccupied | from what you've told me, and from the research i've done myself, it seems the most libre way is literally: linphone or GNU Ring (software applications) connected to SIP provider, with a headset including a mic on it, connected to my laptop (which is a libre-friendly one. libreboot is installed etc) |
| 16:53.59 | [TK]D-Fender | Got just the thing for you then : https://tinyurl.com/bvhslm7 |
| 16:54.25 | unoccupied | i'm not sure if GNU Ring can connect through SIP provider though. i know linphone can. |
| 16:54.54 | unoccupied | [TK]D-Fender: well, that is essentially the same thing as a landline, just less sophisticated |
| 16:55.15 | unoccupied | but yeah that could also work :P |
| 16:55.18 | [TK]D-Fender | If you remove the fan from your CPU you use GNU Smoke. |
| 16:55.51 | unoccupied | yeah and that fan is government by an embedded controller which runs firmware. |
| 16:56.24 | unoccupied | there's no perfect solution yet |
| 16:56.57 | *** join/#asterisk jkroon__ (~jkroon@165.16.204.174) |
| 16:57.12 | [TK]D-Fender | The Del Monte solution above is pretty solid... |
| 16:57.23 | [TK]D-Fender | or .... |
| 16:57.51 | unoccupied | someone i know connects to an asterisk server through their phone, but they took the modem/baseband processor out of it so it can't do GSM etc. isntead, the phone is connected through wifi - from a USB dongle that works with libre driver/firmware, because the phone's own wifi is incompatible without binary blobs - and then talks via SIP protocol |
| 16:58.10 | unoccupied | untrackable wifi-based mobile phone, connected through SIP |
| 16:58.33 | unoccupied | you can only call them if they're connected to a hotspot |
| 16:58.57 | unoccupied | basically what i'm planning to do, but with a laptop |
| 16:59.35 | unoccupied | eh. well, thanks again, and sorry for taking over the channel |
| 16:59.45 | unoccupied | this isn't really related to asterisk at this point |
| 17:01.05 | [TK]D-Fender | Friday is Comedy Night in #asterisk |
| 17:07.48 | *** join/#asterisk iaronson (~luigi@pool-100-15-86-200.washdc.fios.verizon.net) |
| 17:19.32 | *** part/#asterisk iaronson (~luigi@pool-100-15-86-200.washdc.fios.verizon.net) |
| 17:19.55 | *** join/#asterisk luigi86 (~luigi@pool-100-15-86-200.washdc.fios.verizon.net) |
| 17:22.24 | *** join/#asterisk MrMojit0 (~MrMojit0@52D9DF74.cm-11-1d.dynamic.ziggo.nl) |
| 17:39.47 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 17:47.42 | *** join/#asterisk unoccupied (unoccupied@libreboot/developer/leah) |
| 18:12.00 | *** part/#asterisk afournier (~admin@78.233.162.32) |
| 18:17.23 | *** join/#asterisk igcewieling (~ewieling@speedy-02.nyigc.net) |
| 18:27.19 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 18:46.51 | *** join/#asterisk scgm11_ (~scgm11@r186-52-230-215.dialup.adsl.anteldata.net.uy) |
| 18:55.45 | *** join/#asterisk miralin (~Thunderbi@194.8.128.64) |
| 18:59.47 | *** join/#asterisk scgm11__ (~scgm11@2800:a4:176b:d500:e517:4c2c:ca79:9af2) |
| 19:03.50 | *** part/#asterisk Turaiel (~Turaiel@45.55.205.124) |
| 19:27.12 | *** join/#asterisk iBurger21 (~michael@50.3.87.108) |
| 19:30.12 | *** join/#asterisk scgm11_ (~scgm11@2800:a4:176b:d500:e517:4c2c:ca79:9af2) |
| 19:34.28 | *** part/#asterisk iBurger21 (~michael@50.3.87.108) |
| 19:42.59 | *** join/#asterisk Forbidd3n (~forbidd3n@c-98-208-213-159.hsd1.fl.comcast.net) |
| 19:43.57 | Forbidd3n | I am trying to get TLS working with Twilio and asterisk. When it is enabled and Twilio is sending the request I am getting 401 not authorized. |
| 19:43.59 | Forbidd3n | Any suggestions |
| 19:46.36 | [TK]D-Fender | Same as always: Look at the call |
| 19:48.00 | Forbidd3n | In my CLI I keep getting these responses |
| 19:48.04 | Forbidd3n | [2018-03-23 15:47:41] ERROR[58881]: tcptls.c:1095 ast_tcptls_client_start: Unable to connect SIP socket to 54.172.60.0:5061: Connection refused |
| 19:48.04 | Forbidd3n | [2018-03-23 15:47:41] ERROR[58880]: tcptls.c:1095 ast_tcptls_client_start: Unable to connect SIP socket to 54.172.60.1:5061: Connection refused |
| 19:48.16 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 19:48.16 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 19:48.24 | Forbidd3n | those are Twilio's trunk IPs |
| 19:50.07 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 19:51.58 | Samot | Did you setup TLS at all on the PBX? |
| 19:52.39 | Forbidd3n | Samot: yes, I have TLS enabled, I imported our wildcard ssl cert for our server and I added transport=tls to each one of my inbound trunks |
| 19:52.47 | Samot | You cannot use wildcards. |
| 19:52.53 | Samot | I said this the other day |
| 19:52.59 | Samot | It's a rule of TLS over SIP |
| 19:53.19 | Samot | You can't have a cert that is *.domain.com or .domain.com |
| 19:53.39 | Samot | It has to have FDQNs in it. |
| 19:53.40 | Forbidd3n | Ok, would I be using our SSL or should I be using Twilio pem file - https://www.twilio.com/docs/api/sip-trunking/getting-started -- under secure trunking |
| 19:53.46 | Samot | That match. |
| 19:53.57 | Samot | You would be using yours. |
| 19:54.03 | Forbidd3n | Ok, that is what I thought |
| 19:54.10 | Samot | But you cannot use a wildcard certificate |
| 19:54.27 | Forbidd3n | So this would be the reason for the 401 not authorized, correct, if everything works without tls enabled |
| 19:54.32 | Samot | Correct. |
| 19:54.36 | Samot | The cert is wrong |
| 19:54.38 | Forbidd3n | Ok, thanks Samot |
| 19:54.41 | Samot | Invalid. |
| 20:06.31 | Forbidd3n | Thanks Samot. I will get a cert to so FDQN matches |
| 20:08.13 | *** join/#asterisk wearyhacker (~wearyhack@4.13.169.217.in-addr.arpa) |
| 20:30.59 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 20:31.44 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 20:35.07 | jeffspeff | i'm having an odd issue between an asterisk box and a freepbx box. there's an iax trunk configured on each side. the asterisk iax trunk shows that it's up and reachable and i'm able to make calls over it. the iax trunk on freepbx shows as unreachable. i have tried unloading and then loading chan_iax.so but it goes back to 'unreachable'. any suggestions? |
| 20:50.40 | *** part/#asterisk unoccupied (unoccupied@libreboot/developer/leah) |
| 20:54.59 | *** join/#asterisk tzafrir (~tzafrir@62.90.199.247) |
| 21:00.31 | *** join/#asterisk skirmisha (~skirmisha@213.16.62.70) |
| 21:00.41 | skirmisha | hi people |
| 21:01.15 | skirmisha | do you know a way where asterisk sends options/qualify msgs via proxy and not directly? |
| 21:05.14 | skirmisha | anyone? |
| 21:08.37 | *** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il) |
| 21:10.13 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 21:20.09 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 21:48.12 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 22:01.33 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:17.33 | *** join/#asterisk Oooohboy (~Oooohboy@2605:6000:1711:c1ff:38b3:5c5e:423a:7c22) |
| 22:43.34 | *** part/#asterisk kharwell (kharwell@nat/digium/x-spythktrhvtsqufc) |
| 22:46.41 | *** join/#asterisk paulgrmn__ (~paulgrmn@184.75.212.67) |
| 23:22.21 | *** join/#asterisk ih8wndz (jwpierce3@server.maximusabacus.com) |
| 23:37.44 | *** join/#asterisk zigggggy (ffssd@unaffiliated/zigggggy) |
| 23:53.51 | *** join/#asterisk ih8wndz (jwpierce3@server.maximusabacus.com) |