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| 00:19.44 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:19.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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 |
| 00:28.32 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 00:41.04 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 00:43.24 | UncleKiwi | hi there, i am running asterisk 11.24.1 and when i dial my all numbers i get audio. But when i change to 4G ( its a android softphone) all calls except the voicemail work |
| 00:43.29 | UncleKiwi | its strange |
| 00:47.06 | *** join/#asterisk Download-Fritz (~Download-@pD9FE2787.dip0.t-ipconnect.de) |
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| 01:21.40 | UncleKiwi | it seems to be the audio that asterisk plays ie tt-monkeys |
| 01:22.06 | UncleKiwi | but not calls passed through via FXO |
| 01:25.59 | drmessano | Do you have nat=yes set for the extenson? |
| 01:26.03 | UncleKiwi | the calls also look different from the perspective of torch on the router between the two ( asterisk and the softphone ) the ones that have audio bria has a single src port, but the failing ones have two different src ports but same dest ports |
| 01:26.51 | drmessano | Do you have nat=yes set for the extension? |
| 01:26.58 | UncleKiwi | yes i do |
| 01:27.14 | drmessano | What kind of router do you have? |
| 01:27.18 | UncleKiwi | mikrotik |
| 01:28.55 | drmessano | IP > Firewall > Service ports |
| 01:28.56 | drmessano | Disable SIP |
| 01:29.52 | UncleKiwi | yes makes no difference - i ahev played with tha on and off |
| 01:34.59 | UncleKiwi | its strange that there seems to be two streams when it fails |
| 01:35.13 | UncleKiwi | and one bidirectional one when it works |
| 01:36.24 | UncleKiwi | i can see from rtp debug thast asterisk is sending audio > port 26990 and I can see that it is being sent to the soft phone on this port |
| 01:38.03 | UncleKiwi | but the audio in the opposite direction back to asterisk has a different src port than the first stream but same dest port |
| 01:38.14 | drmessano | This is BRIA on Android? |
| 01:39.29 | UncleKiwi | correct |
| 01:39.33 | drmessano | Do you have a Account > Account Advanced > Network Transveral? |
| 01:39.39 | UncleKiwi | si |
| 01:39.40 | UncleKiwi | yes |
| 01:39.47 | drmessano | What is it set to? |
| 01:39.50 | Download-Fritz | does somebody know how to achieve an "session-timers=refuse" equivalent (chan_sip) with chan_pjsip? Even though I have "timers=no", the server still responds to UPDATEs with OK |
| 01:40.36 | UncleKiwi | rport wifi and rport mobile and im using stun mobile and stun for wifi |
| 01:41.01 | drmessano | What about "Outbound Mobile"? |
| 01:41.08 | drmessano | Does that exist? |
| 01:41.21 | UncleKiwi | yes |
| 01:41.23 | UncleKiwi | its off |
| 01:41.44 | UncleKiwi | what does that do ? |
| 01:41.47 | drmessano | I have both RPORTs, Both Outbounds, DNS SRV, and STUN WIFI checked |
| 01:41.53 | drmessano | Rest disabled |
| 01:41.54 | drmessano | and no issues |
| 01:42.12 | UncleKiwi | ok i will now try ... |
| 01:42.13 | drmessano | The Outbounds are the NAT=YES equivalent for the client |
| 01:42.32 | drmessano | and you need to disable that STUN Mobile |
| 01:48.14 | UncleKiwi | mmm same deal |
| 01:48.25 | UncleKiwi | its really odd |
| 01:49.04 | UncleKiwi | because its only the vm and when i get asterisk to play for example tt-monkeys |
| 01:49.33 | UncleKiwi | not calls that are made or recieved via the FXO devices |
| 01:50.03 | UncleKiwi | and if i join any remote wifi it is fine also |
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| 01:50.24 | UncleKiwi | just when using mobile i get this strang fault |
| 01:51.49 | UncleKiwi | whgat version should i be running |
| 01:52.22 | UncleKiwi | 11 or 13 |
| 01:52.35 | UncleKiwi | i have typically been running 11 |
| 01:53.54 | drmessano | Its either your carrier or BRIA settings |
| 01:54.05 | UncleKiwi | ok |
| 01:54.34 | UncleKiwi | it just seems odd that some calls are treated differently |
| 01:54.37 | UncleKiwi | to others |
| 01:55.06 | UncleKiwi | as far at the audio stream |
| 01:55.44 | UncleKiwi | streams/rtp |
| 01:59.26 | UncleKiwi | thank you |
| 02:02.54 | UncleKiwi | drmessano: have you defined your rtp start and end on the bria app ? or is that set to 0 and 0 |
| 02:03.48 | UncleKiwi | advanced settings > media options |
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| 02:34.57 | UncleKiwi | if i enable a dnat rule the audio stream starts ... |
| 02:35.01 | UncleKiwi | how interesting |
| 02:38.10 | UncleKiwi | is there a security risk with having for example 50udp ports dnat'd to the asterisk box ? |
| 02:38.32 | UncleKiwi | eg 10000-10050 |
| 02:38.37 | UncleKiwi | udp |
| 02:39.10 | UncleKiwi | or is that a bad idea |
| 02:40.26 | [TK]D-Fender | You're supposed to be forwarding RTp to your server..... |
| 02:40.27 | [TK]D-Fender | always |
| 02:40.43 | [TK]D-Fender | otherwise they bounce |
| 02:41.48 | UncleKiwi | mmm i have never needed to until i place clients outside of the nat router |
| 02:42.11 | UncleKiwi | i have never needed to open ports if i am registering to a sip provider also |
| 02:43.20 | UncleKiwi | infact just with a dnat to the asterisk box 5061-tcp i was able to get it almost 100% of the time working correct with remote clients |
| 02:43.58 | UncleKiwi | but it seems i need to create a dnat for the rtp ports |
| 02:44.22 | UncleKiwi | you seem to be saying that is normal practice ? |
| 02:45.47 | [TK]D-Fender | <[TK]D-Fender> You're supposed to be forwarding RTp to your server..... |
| 02:45.48 | [TK]D-Fender | <[TK]D-Fender> always |
| 02:45.58 | UncleKiwi | ok thanks |
| 02:46.03 | *** join/#asterisk LiuYan (~NiHola@unaffiliated/liuyan) |
| 02:48.59 | LiuYan | i'm wondering, is there a "station board" which connect analog phone to asterisk, and make analog phone works with asterisk ? This will make us migrating from traditional PBX to asterisk more easily if there's such board existed. |
| 02:49.21 | [TK]D-Fender | SIP Gateway <- |
| 02:49.36 | [TK]D-Fender | yes. Dozens of models to choose from |
| 02:49.47 | [TK]D-Fender | a ATA ... with a lot of ports |
| 02:50.30 | [TK]D-Fender | If it's odd to see a PBX that is using all analog phones as "traditional" |
| 02:51.00 | [TK]D-Fender | Most PBX's used special phones offering special features for the central PBX |
| 02:53.29 | LiuYan | [TK]D-Fender: lol, true. Our PBX brand is Harris which it's company doesn't exist anymore, it DO offering special feature via a 'digital phone' >> odd to see a PBX that is using all analog phones as "traditional" |
| 02:54.04 | LiuYan | [TK]D-Fender: Thanks, I will have a look at that. >> SIP Gateway |
| 02:54.55 | [TK]D-Fender | how many ports do you need? |
| 02:57.02 | LiuYan | [TK]D-Fender: Probably around 100. |
| 02:57.46 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Analog-Gateways |
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| 03:03.41 | LiuYan | [TK]D-Fender: Thanks, Looking at it now, also I found a local manufacture here: http://www.openvox.cn/products/voip-gateways.html |
| 03:04.10 | [TK]D-Fender | Openvox is typically considered cheap junk |
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| 03:07.53 | LiuYan | [TK]D-Fender: Maybe, but price is reasonable, we have several boards of openvox right now, so far no problem (not heavily using though) |
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| 03:12.49 | Download-Fritz | could someone with knowledge about SIP Session Refreshing please have a brief look at this? https://pastebin.com/fqYAhfP0 |
| 03:13.27 | Download-Fritz | Just want to know if the Asterisk server's OK is the correct reaction to the UPDATE. The BYE is likely a provider server screwup, but just want to make sure the Asterisk server is doing fine |
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| 03:51.45 | drmessano | Really? No RTP ports forwarded? |
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| 08:24.32 | polysics | hello! Is anyone running 14 on Docker for Mac? I have a few issues with ports and I'd like to figure out if I am just wasting time :) |
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| 11:19.12 | Martin` | polysics: I'm running in docker on a NAS. If that works it should also run on mac :) |
| 11:19.45 | polysics | Martin`: what network setup do you use? net=host or something else? |
| 11:20.04 | Martin` | in asterisk? |
| 11:20.25 | Martin` | oh on docker |
| 11:20.31 | Martin` | hmm I believe that is what I did |
| 11:23.54 | Martin` | docker instance just uses the host ip :) |
| 11:27.09 | Martin` | I've created a while a ago a config with the gui, now I've no gui anymore, but the config is a mess :P |
| 11:31.01 | Martin` | mybe I go to rewrite the config when I've time, :) |
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| 12:30.56 | Raccoon | MP3 PATENT EXPIRED -- Seen on LAME-dev (mp3 encoder) mailing list. "[Lame-dev] MP3 patents expired; Fraunhofer/Technicolor end licensing program --- http://mp3licensing.com/ now redirects to https://www.iis.fraunhofer.de/en/ff/amm/prod/audiocodec/audiocodecs/mp3.html " |
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| 12:41.52 | polysics | dumb question: does pjsip have a "sip reload"? :D |
| 12:44.19 | Tsunamski | issues.asterisk states the following: module reload res_pjsip.so |
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| 16:51.33 | polysics | hey! does annyone know if it is somehow possible to use streaming TTS with Google Cloud Speech on Asterisk? |
| 16:55.56 | [TK]D-Fender | https://www.google.ca/#q=google+speech+tts+asterisk |
| 16:57.12 | [TK]D-Fender | Never ask a question you can have Google answer instantly with 1/3 the # of words... |
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| 17:10.36 | igcewieling1 | Sorry, Asterisk, I've found a new love: Arduino 8-) |
| 17:29.07 | lorsungcu | alright cool |
| 17:29.36 | polysics | [TK]D-Fender: "streaming" is the keyword there, ie. using hte chunked transfer |
| 17:29.42 | polysics | which Google might not even offer |
| 17:29.52 | polysics | looking at it, I am not even sure Google has an API :D |
| 17:29.59 | polysics | so s/Google/Watson/ |
| 17:31.09 | polysics | though I am not even sure streaming matters - those APIs are pretty fast |
| 17:31.23 | polysics | but I think streaming would be audibly better |
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| 17:48.04 | polysics | I will also ask another question I had earlier: is anyone running Asterisk inside a Docker container on Docker for Mac? |
| 17:48.10 | polysics | it looks like it has a lot of issues |
| 18:18.21 | dan_j | Hi. If a callee uses a dymanic feature which uses agi to transfer a caller, is it possible to maintain the callee's channel and continue the dynamic feature dialplan? |
| 18:19.11 | dan_j | Or does the Transfer application always terminate the caller's channel? |
| 18:26.17 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:28.28 | sawgood | Hi: question: using Asterisk 13: using FXO lines (analog): When a call arrives to the Asterisk PBX from the FXO line (and) rings the phone (but there is no answer) and the call desired to then leave back out the PBX (call forwarding) (to the cell of the staff member) ..... is it at all possible to script Asterisk (to make that 2nd) call out (using the same) phone line as the line the call |
| 18:28.28 | sawgood | came in on (with something like) "3-way calling"? |
| 18:29.48 | sawgood | go back out (for) call forwarding (on the same) FXO channel as it "came in" on (with) the phone company service called "call waiting" or "3-way calling" (I ask this) because I know of a PBX that does do this ... (might be the only one) one the market that does??? |
| 18:33.22 | sawgood | TalkSwitch PBX (now Fortinet) did this with their patented servcie called, "same line connect" (if you had) only 1 FXO in your PBX (you could) make two phone calls .... (without) the telco servcie of "3-way calling" |
| 18:39.36 | [TK]D-Fender | if you use an interface you can flash to do this with, yes |
| 18:39.45 | [TK]D-Fender | DAHDI can. |
| 18:40.01 | [TK]D-Fender | I've never heard of any SIP gateway offering it |
| 18:47.39 | sawgood | thanks ... FLASH key (on an older) analog phone (I've done that) .... |
| 18:51.22 | sawgood | back in the day: when SIP was much more rare: people had early on IP PBX units which took FXO lines only (and) sometimes only had 1 or 2 lines in their system for everyone to share .... |
| 18:52.07 | sawgood | having the PBX use the same line to make that 2nd call out (helped) keep the system open for incoming calls, etc .... (I'm going) to ask the customer to consider a digital trunk because their phones are SIP too ... |
| 18:52.54 | [TK]D-Fender | that doesn't matter |
| 18:53.03 | [TK]D-Fender | changing lines is for line's sake |
| 18:53.31 | [TK]D-Fender | not just because local handsets are SIP |
| 18:53.38 | sawgood | right on ... |
| 19:08.01 | *** join/#asterisk matt_ (~matt@ccpc-buzzer.bath.ac.uk) |
| 19:08.21 | matt_ | does anybody know if its possiable to change sip settings in the dialplan ? |
| 19:08.27 | matt_ | for just that channel |
| 19:09.01 | file | ...like what? |
| 19:09.03 | sawgood | matt_: do you mean in sip.conf? |
| 19:10.00 | matt_ | sawgood: no in extensions.conf |
| 19:10.34 | matt_ | sawgood: so I have canreinvite=no set in [global] but I would like to set it to =yes for a single exten |
| 19:11.24 | file | you have canreinvite=no... where... |
| 19:11.34 | matt_ | file: in sip.conf [global] |
| 19:11.37 | sawgood | maybe if you had that extension in its own context in (sip.conf) |
| 19:12.03 | file | that's the only way. |
| 19:12.08 | matt_ | sawgood: theres no authentication on this machine, its setup to just accept calls from the internal and deliver them to internal hosts |
| 19:12.19 | matt_ | *from the internet |
| 19:12.36 | matt_ | file: humm, ok thanks |
| 19:15.32 | matt_ | humm, unless I can set the networks with hosts I want to stay inline for as localnet='s and say only reinvite if the destination host isn't in the localnet's list? |
| 19:16.00 | file | I don't believe the reinvite code checks that |
| 19:16.21 | sawgood | sorry I don't know ... |
| 19:16.26 | matt_ | ok cheers |
| 19:18.02 | matt_ | maybe create another machine on a different public ip that always reinvites, and just let it reinvite to the existing machine if it needs to |
| 19:19.40 | *** join/#asterisk asteriskmonkey (~philip@162.250.193.47) |
| 19:19.59 | asteriskmonkey | heya, been a while since i used this channel :P |
| 19:20.16 | asteriskmonkey | is there a dialplan syntax checker out yet for vi/notepad++ etc? |
| 19:20.53 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 19:21.02 | [TK]D-Fender | vi yes IIRC |
| 19:21.26 | asteriskmonkey | [TK]D-Fender: you have link for formatted? |
| 19:21.42 | [TK]D-Fender | https://www.google.ca/#q=asterisk+vi+syntax |
| 19:22.01 | [TK]D-Fender | Feel free to google for others |
| 19:22.04 | asteriskmonkey | ah sweet thanks |
| 19:23.56 | TheGallopingFox | does asterisk win awards for having the most amount of config files for a single application? :P |
| 19:24.36 | file | we could take vast amounts of functionality away... |
| 19:24.49 | TheGallopingFox | no, leave them :) |
| 19:24.58 | matt_ | lol, i start with no configs and just add the ones I need |
| 19:25.17 | asteriskmonkey | what? |
| 19:25.19 | asteriskmonkey | you already do |
| 19:25.31 | TheGallopingFox | i disabled a few things in modules.conf then just added all the configs, there are no errors or warnings in my log |
| 19:25.32 | asteriskmonkey | any large scale operation is opensips/karmailio doing most of work lol |
| 19:25.50 | asteriskmonkey | ah yeah its got alot of kit in default build :) |
| 19:26.04 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 19:26.21 | TheGallopingFox | i think its nice having all the configs in place so you have a idea what asterisk can do |
| 19:27.35 | TheGallopingFox | i just did this /etc/asterisk ls -1 | wc -l |
| 19:27.41 | TheGallopingFox | 111 configs!!!! |
| 19:27.42 | *** join/#asterisk miralin (~Thunderbi@194.8.128.114) |
| 19:28.33 | TheGallopingFox | im pretty sure asterisk can do just about anything :) |
| 19:28.36 | polysics | I think a superminimal config is asterisk.conf, modules.conf, logger.conf (oddly does not boot otherwise), extensions.conf and your channel driver(s) like pjsip.conf |
| 19:28.52 | matt_ | TheGallopingFox: cool, i got 20 :) and you dont need the -1 |
| 19:29.24 | igcewieling1 | TheGallopingFox: on my systems there are 31 config files in /etc/asterisk |
| 19:29.34 | TheGallopingFox | to have a super minimal configs you need to disable a lot of stuff in modules.conf to stop errors and warnings in the logs |
| 19:29.40 | drmessano | I would prefer if Asterisk had one 4MB config file |
| 19:29.47 | drmessano | Easy to scroll |
| 19:29.50 | drmessano | Endlessly |
| 19:29.58 | matt_ | polysics: is pjsip better than standard asterisk sip? ive never use it |
| 19:30.36 | drmessano | TheGallopingFox: You dont have to compile or autoload ANY of those modules |
| 19:30.44 | drmessano | So basically, you're doing it wrong |
| 19:31.01 | polysics | matt_: there are many reasons why they switched, mostly regarding the fact that chan_sip has become unmaintainable |
| 19:31.09 | drmessano | Only load the modules you need = Few config files |
| 19:31.12 | TheGallopingFox | i do have this turned on: autoload=yes |
| 19:31.17 | drmessano | Yep |
| 19:31.21 | drmessano | That loads ALL |
| 19:31.26 | TheGallopingFox | ohhh hell |
| 19:31.38 | drmessano | Everything you've crapped into the modules directory during install |
| 19:31.43 | polysics | I would not say pjsip is "better" |
| 19:31.45 | drmessano | Like MGCP and other horrible things |
| 19:32.12 | polysics | how do you load JUST pjsip and manager modules? pjsip is composed of a few things |
| 19:32.13 | drmessano | So basically, "What do you think will happen if you compile ALL modules and then load ALL of them?" |
| 19:32.18 | [TK]D-Fender | In most ways it is better |
| 19:32.23 | igcewieling1 | I don't know if pjsip is better, but I know the documentation for pjsip with Asterisk is a lot worse than docs for chan_sip. |
| 19:32.34 | matt_ | polysics: ok cool, might take a look |
| 19:32.46 | [TK]D-Fender | Supports multiple registrations, multiple contacts, multiple transport bindings, etc |
| 19:32.50 | polysics | ok, so, "I would not say" is wrong, "I do not know if it is better but experts know and they say it is" |
| 19:33.19 | polysics | which one of the pjsip_* modules is "just pjsip"? |
| 19:33.26 | igcewieling1 | my modules.conf https://pastebin.com/i0sXyG79 |
| 19:33.42 | file | igcewieling1: can you clarify how it is worse? just that chan_sip has more? |
| 19:33.46 | igcewieling1 | I use Asterisk 11, of course. |
| 19:33.55 | polysics | you have to use noload? |
| 19:34.02 | polysics | oh, with autoload |
| 19:34.05 | *** join/#asterisk Micc (~micster@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 19:34.09 | polysics | isn't it faster to not autoload? :D |
| 19:34.10 | igcewieling1 | file: There is a lot more documentation and community knowledge for chan_sip. |
| 19:34.21 | Micc | I've just upgraded my server to asterisk 14.3.0 from 13.7.2 and I'm getting a strange error whenever I'm using mixmonitor to record a call. Unable to translate to format wav49, source format slin |
| 19:34.27 | file | k |
| 19:34.33 | TheGallopingFox | do i need to specify load => for each module i require instead of load all ? |
| 19:34.52 | drmessano | There's a lot of INCORRECT documentation for chan_sip too |
| 19:34.57 | Micc | the call comes in using ulaw which I'm guessing asterisk converts to slin or something internally. |
| 19:35.00 | matt_ | Micc: i mixmonitor to alaw but then all my calls are alaw |
| 19:35.02 | drmessano | I would say 75% of it is old and outdated |
| 19:35.16 | drmessano | and more like 99% of chan_pjsip documentation is actually correct |
| 19:35.23 | drmessano | So quality vs quantity |
| 19:35.33 | polysics | anything on voip-info looks wrong. |
| 19:35.38 | Micc | matt_, this used to work fine in asterisk 13. I feel like I must be missing a module or something. |
| 19:35.45 | drmessano | everything on voip-info is supect |
| 19:36.06 | drmessano | They've done an AMAZING job with the Asterisk Official docs |
| 19:36.11 | Micc | I really like recording into WAV since it's small and easy to convert to mp3 on the back end. I don't really want to have to change all of that. |
| 19:36.20 | drmessano | Few reasons to look past those to other sources |
| 19:36.21 | matt_ | Micc: maybe, not sure tbh, maybe not linked against a library? try writing to a filename of something.ulaw and see what happens |
| 19:36.27 | drmessano | ESPECIALLY with PJSIP |
| 19:36.57 | drmessano | Sure, there are more and varied EXAMPLES other places |
| 19:37.02 | matt_ | humm, did I just start a flame war :/ |
| 19:37.05 | matt_ | runs away |
| 19:37.10 | drmessano | But I use the Asterisk Docs when I want to be CORRECT about usage |
| 19:37.28 | asteriskmonkey | and i though opus was still the trigger word :P |
| 19:37.43 | drmessano | Nobody is arguing about anything |
| 19:38.01 | matt_ | drmessano: are you sure? |
| 19:38.12 | drmessano | I don't see anyone arguing |
| 19:38.19 | polysics | asteriskmonkey: WHO SAID OPUS??? |
| 19:38.40 | TheGallopingFox | is there a simple way to find out what config requires what module? |
| 19:38.51 | drmessano | I made a statement about pjsip docs being less prolific, but actually correct in almost all cases |
| 19:39.14 | drmessano | and chan_sip docs being much more widespread, but mostly outdated |
| 19:39.34 | drmessano | Facts? |
| 19:40.19 | asteriskmonkey | why didnt iax make it big :( |
| 19:40.28 | drmessano | Because it wasn't needed |
| 19:40.37 | drmessano | chan_sip got much better |
| 19:40.51 | drmessano | So people stopped using IAX as an excuse |
| 19:40.59 | Micc | ok, so I can record to .wav but not to .WAV |
| 19:41.17 | Micc | am I missing something like format_wav49? |
| 19:41.27 | drmessano | Asterisk 1.6.x.x put the nails in IAX2's coffin.. So much chan_sip work |
| 19:41.38 | Micc | yes, yes I am. |
| 19:42.02 | Micc | no, there is no such thing. shit. |
| 19:43.23 | Micc | I still have a couple customers that prefer using IAX |
| 19:43.34 | igcewieling1 | IAX didn't make it big because everyone uses SIP and nobody added support for IAX to their products, except Zoiper |
| 19:43.44 | matt_ | Micc: humm, I didn't think .WAV was a different format to .wav tbh |
| 19:44.00 | igcewieling1 | If polycom or Cisco supported iax, I suspect it would be far more popular. |
| 19:44.07 | matt_ | igcewieling1: I use IAX, i use it internally but I use SIP for internet things |
| 19:44.17 | Micc | matt_, wav is uncompressed I believe and WAV is compressed. in asterisk the formats are wav and wav49 |
| 19:44.29 | matt_ | Micc: ahh ok |
| 19:44.59 | igcewieling1 | matt_: a number of years ago I had significant issues with IAX, since it provided me with nothing that SIP could not provide, I stopped using it. |
| 19:46.10 | Micc | I think I see. I compared core show translation on two different servers. I'm missing gsm on the new server. I think it needs it for some reason. |
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| 19:48.47 | matt_ | igcewieling1: ok, yea I dont think iax gives my anything different, it used to be nice for nat's but theres a tunnel between all the machines I run these days |
| 19:48.49 | drmessano | igcewieling1: There were other implementations.. But that didn't matter. chan_sip simply got better |
| 19:48.58 | drmessano | So for Asterisk, nobody needed it |
| 19:49.43 | drmessano | TCP/TLS/SRTP, stability, speed... lots of great improvements in just a few years |
| 19:50.47 | drmessano | IAX was the "Anti-SIP" for pretty much anyone that was asked |
| 19:52.25 | Samot | â¤ï¸ SIP |
| 19:53.36 | matt_ | Micc: oh yea, the both work for me |
| 19:54.05 | matt_ | Micc: test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 19:54.14 | matt_ | Micc: test1.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz |
| 19:54.37 | matt_ | looks like WAV requires gsm and wav requires pcm |
| 19:56.15 | Micc | matt_, yup. BTW, what do you use to tunnel between machines? |
| 19:56.23 | matt_ | Micc: openvpn |
| 19:56.49 | matt_ | Micc: well, i have one link thats tinc |
| 19:57.09 | Micc | matt_, do you use bridge or nat? |
| 19:57.52 | matt_ | I was trying its clustering multi peering stuff but its easier for me to just create p2p tun's all over the place and use bgp to get traffic going in the right direction |
| 19:58.05 | matt_ | Micc: erm.. neither |
| 19:58.56 | matt_ | routing, and the hosts all have /32 bound to their loopbacks that they advertise, so if a link between two machine goes down traffic can route through a third if needed |
| 19:59.40 | Micc | matt_, that sounds badass. I need to setup something like that for my servers. |
| 20:00.48 | matt_ | Micc: ive found it to be the best way if you have like a cable line and adsl line to a site, you can create two tunnels over each link and get bgp to always use the cable line unless its not there |
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