| 00:18.47 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 00:18.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0-rc1 (2018/07/03), Standard: 15.5.0-rc1 (2018/07/03); 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 |
| 00:31.12 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
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| 01:10.49 | *** join/#asterisk flexnsniff (~sparky@173-28-0-255.client.mchsi.com) |
| 01:11.38 | flexnsniff | so... any of you folks have experience with asterisk-to-asterisk trunking with PJSIP? |
| 01:33.03 | tuxd00d | Yes, most of us. |
| 01:39.46 | flexnsniff | I can't seem to get it to work. I've got pretty much everything else working after an upgrade to PJSIP... dahdi, sccp, SIP trunks to VoIP.ms and Vitelity... |
| 01:40.32 | flexnsniff | TLS/SRTP endpoints too, but still can't figure out how to properly authenticate SIP going asterisk to asterisk |
| 01:41.05 | tuxd00d | Using IP auth, correct? |
| 01:41.20 | flexnsniff | Nope, one side is dynamic IP so can't do it both ways |
| 01:43.25 | flexnsniff | I've got it's FQDN in there, dynamic dns has worked for me so far. but thats not IP auth, just NAT info... |
| 01:44.14 | tuxd00d | Packets are showing up on both ends? |
| 01:46.58 | flexnsniff | Yeah, they've got IP connectivity... I get logs on both sides during an error |
| 01:47.05 | flexnsniff | Putting together config/logs right now |
| 01:47.39 | flexnsniff | It's got to be user error... I don't know enough about PJSIP configuration |
| 01:52.16 | tuxd00d | Is there a reason you need two Asterisk talking to each other? |
| 01:54.43 | flexnsniff | Cloud box runs low-latency, low-jitter on AWS, so my buddy and I can have OPUS calls over VoIP. |
| 01:54.51 | flexnsniff | Works like shit on my cable or DSL connection |
| 01:55.28 | flexnsniff | My home box is part of my homelab, and has FXO/FXS, Skinny, and whatever else |
| 01:57.09 | flexnsniff | https://pastebin.com/5fve8q86 |
| 01:57.38 | flexnsniff | imporant pjsip.conf sections of both sides, pjsip log of one side during restart of the other |
| 01:59.23 | flexnsniff | On top of all this, once I get it working, I'm trying to force it to TLS/SRTP |
| 01:59.47 | tuxd00d | raises the question: "Why would anyone want to hear me more clearly?" |
| 02:00.08 | flexnsniff | I had it working with chan_sip but, no, I decided I wanted to "catch up with the times" and make sure I had a "patched and upgraded system" /s |
| 02:00.47 | flexnsniff | 48khz voice traffic is awesome. I can hear what the other person is watching on TV. "Can you hear this right now?" "I sure can." |
| 02:00.55 | Nivex | http://docs.polycom.com/global/documents/whitepapers/effect_of_bandwidth_on_speech_intelligibility_2.pdf |
| 02:01.26 | Nivex | <4KHz audio makes me cringe now |
| 02:02.11 | tuxd00d | So your issues is with connecting your home Asterisk to your AWS Asterisk? |
| 02:02.21 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 02:03.22 | tuxd00d | Without seems some actually error messages, it's hard to say what's happening. |
| 02:05.57 | flexnsniff | both ways. home box says something lole failed to authenticate 401 |
| 02:07.27 | flexnsniff | *like... |
| 02:07.45 | tuxd00d | The config to your ITSP should be essentially identical to your config to connect to your AWS box. |
| 02:08.19 | flexnsniff | which is what i thought too, my voip.ms account looks nearly the same |
| 02:09.04 | tuxd00d | Your VOIP.ms is using user/pass auth? |
| 02:09.22 | flexnsniff | same thing for endpoints, i have others configured that work |
| 02:09.29 | flexnsniff | so should be close to the same |
| 02:09.33 | flexnsniff | yep. |
| 02:09.38 | tuxd00d | Before I get ahead of myself. Your ITSP accounts are on the AWS box or your home box? |
| 02:09.59 | flexnsniff | both. two on one one on another |
| 02:10.18 | flexnsniff | verified working within their own syatem |
| 02:10.35 | tuxd00d | ~pb |
| 02:10.35 | infobot | it has been said that 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. |
| 02:11.08 | tuxd00d | actually, I think those are all broken. pastebin.com should work. |
| 02:12.05 | tuxd00d | you need to show the packets. |
| 02:13.20 | tuxd00d | But it *sounds* like you simple don't have matching auth credentials, forgot to reload credentials, .... |
| 02:15.47 | flexnsniff | hmmmm okay |
| 02:16.01 | flexnsniff | I can use my own website to post a .pcap lol |
| 02:16.08 | tuxd00d | However, if you have latency issues connecting through your home box, you're going to have latency issues connecting through AWS from home. |
| 02:16.22 | flexnsniff | just need to figure out how to do it... two headless servers... |
| 02:16.46 | flexnsniff | Nah I don't have issues, just my buddy in arizona does. |
| 02:16.56 | tuxd00d | Servers have heads? .... must be a Microsoft thing... |
| 02:17.10 | flexnsniff | =P |
| 02:17.41 | tuxd00d | He must have CenturyLink. |
| 02:18.13 | flexnsniff | Nah, just me =P |
| 02:18.41 | flexnsniff | The jitter is low in my area, but it seems it's internet routing has worse jitter than my cable connection |
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| 02:20.02 | flexnsniff | seems like the connection between servers provides an extra layer of buffer |
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| 02:33.29 | flexnsniff | well, I captured it, but it's just as worthless as what I've got now |
| 02:33.44 | flexnsniff | two ideas: A.) its not sending auth to begin with |
| 02:34.01 | flexnsniff | B.) I see a s@example.com and i think it got truncated somehow... gotta check that out |
| 02:39.15 | flexnsniff | FOR THOSE of you who were also confused: the "contact" sip URI was wrong. edited to sip:user@example.com and fixed it. |
| 02:39.21 | flexnsniff | user error..... |
| 02:40.17 | *** join/#asterisk tuxd00d (~tuxd00d@unaffiliated/tuxd00d) |
| 02:41.03 | tuxd00d | 's' is start .. when a DID isn't sent with the INVITE. |
| 02:41.23 | tuxd00d | or something to that effect.... |
| 02:48.14 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions |
| 02:49.11 | flexnsniff | yeah I thought I had seen that before but the contact URI was in fact the problem for registration |
| 02:49.34 | flexnsniff | now my issue is the dialplan... passing SIP over a SIP isn't working like I previously expected |
| 02:50.33 | flexnsniff | used to be Dial(SIP/trunk/${EXTEN}) |
| 02:53.17 | [TK]D-Fender | "passing SIP over a SIP isn't working" <- huh? |
| 02:57.09 | flexnsniff | I know right... SIP trunk between two asterisk boxes, passing an internal SIP call over the trunk |
| 02:57.24 | [TK]D-Fender | No such thing as "internal |
| 02:57.27 | [TK]D-Fender | a SIP call is a SIP call |
| 02:57.50 | flexnsniff | okay, so how do I do this then, smartypants? |
| 02:57.51 | [TK]D-Fender | From a phone to * from * to another *. From * to another phone |
| 02:57.55 | [TK]D-Fender | all separate |
| 02:58.03 | [TK]D-Fender | you need to be clear on which leg you're referring to |
| 02:58.07 | flexnsniff | but its a sip account on a different server |
| 02:58.29 | [TK]D-Fender | Whcih you define on your own. |
| 02:59.09 | [TK]D-Fender | <flexnsniff> used to be Dial(SIP/trunk/${EXTEN}) <--- like [trunk] you're referring to here |
| 02:59.26 | flexnsniff | It used to be Dial(SIP/1000) for a "local" account or Dial(SIP/trunk/2000) for a "remote" account "2000" over the trunk "trunk" |
| 02:59.44 | [TK]D-Fender | no |
| 02:59.48 | [TK]D-Fender | that is not how it works at all |
| 03:00.00 | [TK]D-Fender | you dia a NUMBEr on the other server you don't dial a direct peer ever |
| 03:01.11 | [TK]D-Fender | they peer you dial is the authed account between your servers, nothing to do with [2000] on the receiving end. When your call is accepted their dialplan is responsible for handling what you dial |
| 03:02.10 | flexnsniff | of course, its the other server's dialplan that handles where the '2000' goes to |
| 03:02.53 | [TK]D-Fender | Your previous phrasing said something very different |
| 03:02.56 | flexnsniff | what I'm asking though, is the syntax between SIP and PJSIP, how do I call a number at a server |
| 03:03.02 | [TK]D-Fender | So, where does that leave us? |
| 03:03.27 | [TK]D-Fender | PJSIP... |
| 03:04.34 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 03:05.22 | flexnsniff | yeah, reading that now. Seems like Dial(PJSIP/${EXTEN}@mytrunk) should work. I think its my other server's dialplan that isn't picking it up anymore |
| 03:05.32 | Nivex | <PROTECTED> |
| 03:05.34 | *** part/#asterisk Nivex (nivex@triton.nivex.net) |
| 03:05.54 | [TK]D-Fender | You should see this very explicitly... |
| 03:06.31 | flexnsniff | I tried the ${EXTEN:1}, didn't realize that just matches the XXX of a _7XXX, but i can see on the far server side that it can't figure out where to pass a 3-digit call |
| 03:07.14 | [TK]D-Fender | You control both servers, no? |
| 03:08.41 | flexnsniff | Yes. |
| 03:09.05 | flexnsniff | I think I'm just running out of useful conciousness |
| 03:09.10 | [TK]D-Fender | I supposed you should know how that dialplan was setup then so that you have matches where you are sending the call |
| 03:09.45 | [TK]D-Fender | <flexnsniff> I think I'm just running out of useful conciousness <- that's an important thing alright |
| 03:09.55 | flexnsniff | Well, I did have this all working... then I upgraded to PJSIP on the same Ast13 |
| 03:10.06 | flexnsniff | moved my config files over without editing much |
| 03:10.13 | flexnsniff | but most everything broke .... |
| 03:11.59 | flexnsniff | I think it has to do with PJSIP and aor/contact points |
| 03:12.43 | flexnsniff | once the call has passed through, it knows what user its supposed to reach, but then the aors/contact lookup isn't working after incoming |
| 03:14.15 | [TK]D-Fender | So your dialpla is processing and it's trying to Dial() on that receiving end's dialplan? |
| 03:22.46 | flexnsniff | well, its supposed to |
| 03:23.19 | flexnsniff | i see a global variable getting changed right as it goes theough on the receiving wnd |
| 03:23.39 | flexnsniff | going to investigate that after this beer |
| 03:30.39 | [TK]D-Fender | That doesn't sound like you're looking at CLI afor this call... |
| 03:31.07 | [TK]D-Fender | it's either landing where it should and getting to a Dial() ... or it isn't |
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| 03:34.12 | *** part/#asterisk chiggins (~chiggins@unaffiliated/chiggins) |
| 03:47.13 | flexnsniff | "local" side == Everyone is busy/congested at this time (1:0/0/1) |
| 03:47.28 | flexnsniff | "remote" side == Setting global variable 'SIPDOMAIN' to 'dsl.sparxdsm.com' |
| 03:48.04 | flexnsniff | that's all I freakin get dude. I know, it should work or not work, or give me something more than THAT. |
| 03:52.44 | [TK]D-Fender | So far nothing tells me the call was accepted |
| 03:52.58 | [TK]D-Fender | and you aren't looking at SIP debug and showing us the call arriving |
| 03:53.02 | [TK]D-Fender | ~pb |
| 03:53.03 | infobot | i guess 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. |
| 03:53.04 | [TK]D-Fender | ^^^^ |
| 03:58.29 | flexnsniff | [TK]D-Fender> So far nothing tells me the call was accepted |
| 03:58.34 | flexnsniff | This was right. |
| 03:59.02 | flexnsniff | 488/Not Acceptable Here |
| 03:59.48 | flexnsniff | SIP is working just fine, codec/media transport negotiation is not |
| 04:02.07 | [TK]D-Fender | then SIP is NOT working |
| 04:02.13 | [TK]D-Fender | working means the call made it |
| 04:02.16 | [TK]D-Fender | that is a failure |
| 04:02.36 | [TK]D-Fender | So get that sorted and we'll see what's next |
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| 04:11.00 | flexnsniff | Hmm, invite>401>ack, invite>488>trying>ack |
| 04:11.37 | flexnsniff | 401 is the error i previously got during registration woes... so mess with the URI? (thats what fixed that)... |
| 04:11.41 | flexnsniff | idk, im at a loss. |
| 04:11.43 | flexnsniff | of sleep. |
| 04:12.22 | flexnsniff | And this is on the receiving end |
| 04:12.26 | [TK]D-Fender | You keep asking what to do and we've seen absoultely nothing |
| 04:12.33 | [TK]D-Fender | we hav no idea what the actual debug looks like |
| 04:12.45 | [TK]D-Fender | Or if your interpretation of it is correct. |
| 04:12.55 | [TK]D-Fender | Or the configs that it's going through |
| 04:13.16 | flexnsniff | Yeah, giving up more security for a debug that's not really gonna help |
| 04:13.32 | [TK]D-Fender | It would |
| 04:13.49 | [TK]D-Fender | because you're expecting to get advice on fixing a car when we can't even see the motor |
| 04:14.01 | [TK]D-Fender | Want an autopsy? Give us a body. |
| 04:14.07 | flexnsniff | Meh, thats a bad analogy |
| 04:14.13 | [TK]D-Fender | Your call is dead |
| 04:14.17 | [TK]D-Fender | It's a perfect analogy |
| 04:14.22 | flexnsniff | my dad has helped me fix a car over the phone from 1k miles away |
| 04:14.34 | [TK]D-Fender | Trusting some kind of feedback from you |
| 04:14.41 | [TK]D-Fender | and you're spitting out tiny tidbitys |
| 04:14.46 | [TK]D-Fender | and it isn't cutting it |
| 04:15.02 | [TK]D-Fender | We can' |
| 04:15.08 | [TK]D-Fender | t be sure of what's wrong |
| 04:15.14 | flexnsniff | This is why I dont do IRC. Thanks for your help. |
| 04:15.19 | [TK]D-Fender | with what you've given. |
| 04:15.24 | flexnsniff | I appreiciate it. Youguys got me pretty far as it is |
| 04:15.27 | flexnsniff | with the "tiny bits" |
| 04:15.28 | [TK]D-Fender | We know how to read debug |
| 04:15.37 | flexnsniff | I only want a hand, not a wheelchair |
| 04:15.37 | [TK]D-Fender | And see where the problem is. |
| 04:15.50 | [TK]D-Fender | But if we're not going to see anything... then we aren't going to run guessing at things either |
| 04:16.22 | flexnsniff | The error is in the 401. At the receiving side. Why? I don't know. |
| 04:16.30 | [TK]D-Fender | You said you got a 4888 |
| 04:16.35 | flexnsniff | I'm offering the same codecs |
| 04:16.38 | flexnsniff | AFTER the 401 |
| 04:16.40 | [TK]D-Fender | If you get that after then the 401 means nothing |
| 04:17.08 | flexnsniff | So the 488 then? Codec mismatch? My configs are the same!?! |
| 04:17.24 | *** join/#asterisk miralin (~Thunderbi@194.8.128.63) |
| 04:17.33 | [TK]D-Fender | Codec, or other protocl bit like encryption, etc |
| 04:17.37 | flexnsniff | Only other 488 i've gotten is SRTP mismatch |
| 04:17.40 | flexnsniff | OOOOOOOH |
| 04:17.51 | [TK]D-Fender | the MEDIA you want to pass isn't allowed |
| 04:17.58 | flexnsniff | My endpoints are TLS/SRTP, the link between asterisk boxes is not |
| 04:18.00 | [TK]D-Fender | What & HOW |
| 04:18.02 | flexnsniff | Is that?! |
| 04:18.06 | [TK]D-Fender | is it? |
| 04:18.14 | [TK]D-Fender | it's YOUR config. |
| 04:18.18 | flexnsniff | I freakin bet it is |
| 04:18.26 | [TK]D-Fender | if we saw that call we could see what was offered |
| 04:18.40 | [TK]D-Fender | and twhat's supported by the peer it's supposedly matching |
| 04:18.43 | flexnsniff | Yeah, and sanitizing 20+ passwords and 100's of IP's is a biznotch |
| 04:18.45 | [TK]D-Fender | It's all got to agree |
| 04:18.58 | flexnsniff | I have other peoples privacy to look after too, not just my own |
| 04:19.01 | [TK]D-Fender | you don't have 20 passwords on the ONE section we need |
| 04:19.13 | [TK]D-Fender | And it isn't shoved in plain text in the SIP debug either |
| 04:19.18 | flexnsniff | Well I thought I was giving appropriate sections |
| 04:19.24 | flexnsniff | but they were "tiny" =P |
| 04:19.45 | flexnsniff | tell me exactly what you need or get what you're given (are you my wife?!) |
| 04:21.26 | [TK]D-Fender | pastebin the complete attempt from beginning to end |
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| 04:59.06 | flexnsniff | Working now. Changed trunk to TLS just to be sure, but that didnt do anything |
| 04:59.36 | flexnsniff | Specificially set codecs on both client and server, both sides (client and server run on both) |
| 05:00.02 | flexnsniff | works like a champ now. Only time i've ever encountered a 488 was SRTP or codec problems. I should have know to trust my own experience |
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| 05:13.33 | flexnsniff | sweet, got dahdi & sccp fixed. Trunking working. Woot. |
| 05:14.22 | drmessano | Oh dear god |
| 05:14.37 | drmessano | You typed SCCP and I woke from a dead sleep |
| 05:15.00 | flexnsniff | now another question: anyone have experience with LetsEncrypt and asterisk?? =) |
| 05:15.19 | drmessano | What is there to experience? It's just certs |
| 05:15.24 | flexnsniff | yeah, SCCP was a nightmare for me too... professor at school gave me a cisco 7910, said it "didnt work" |
| 05:15.54 | flexnsniff | that was a fun experience. It now works and sits on my rack |
| 05:16.32 | flexnsniff | Automatically renewing and keeping asterisk pointing towards it.. Just a matter of fiddling with permissions? |
| 05:17.13 | drmessano | Just be sure the replacement certs are readable by Asterisk. Done. Like any other SSL cert |
| 05:17.20 | flexnsniff | i assume "certbot --certonly" and then point asterisk to where you shit them out at? |
| 05:17.24 | drmessano | If you renew the cert, it has to be readable |
| 05:17.50 | flexnsniff | There lies the problem. That's more cron/script than asterisk or SSL |
| 05:18.16 | flexnsniff | Well I've got enough done for the night, that gives me something to break tomorrow |
| 05:18.34 | drmessano | You're right, it's not an Asterisk issue at all. You renew the cert, chown it properly |
| 05:19.31 | flexnsniff | can I just make asterisk a member of the group the cert belongs to? that would work too right? |
| 05:19.47 | drmessano | That should be fine |
| 05:20.04 | drmessano | There's no magic here involving Asterisk |
| 05:20.06 | flexnsniff | or is it picky like SSH is, must be chmod "400" |
| 05:20.12 | drmessano | The asterisk user needs to be able to read it |
| 05:20.13 | drmessano | Done |
| 05:20.35 | flexnsniff | well there could be, like i mentioned for ssh. It checks to make sure the permissions aren't too permissive |
| 05:20.45 | drmessano | Im telling you there is NOT |
| 05:20.53 | flexnsniff | But you're telling me no, asterisk is dumb, it doesnt give a F, just reads the damn file |
| 05:20.56 | flexnsniff | k. |
| 05:21.23 | drmessano | Como esta Asterisk el reado file permissio gordita SSL |
| 05:21.37 | drmessano | Ez Pz |
| 05:26.09 | flexnsniff | "el easy" |
| 05:26.28 | flexnsniff | nothing like certs in vCenter Server Appliance then, eh? |
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| 08:06.38 | _fuz | Hi. I want to set up a new asterisk with sip trunking. Does anybody has experience using easybell or einfachvoip as sip providers? |
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| 08:29.29 | _fuz | Taking about chan_sip vs chan_pjsip. Is chan_pjsip stable? What do you prefer and why? I just know chan_sip. Does it make sense to learn pjsip? |
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| 13:22.08 | pawiecki | runing a simple script (asterisk -x "sip show peers") locally works fine, but when I try to run it remotely cia web browser gives me errorUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) What can be the problem? |
| 13:23.15 | pawiecki | ah now I see it, permissions for the asterisk.ctl socket are wrong |
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| 13:43.09 | rfr__ | Hello All, I have an old Zaptel dial plan ( that works well on old Zaptel/ * servers ) that I am trying to convert to work on a new Dhadi / * server. So far, I am unable to make outbound calls -- the dial tone continues after I've picked up the phone and started dialing. Anyone have an idea what might be wrong? Here is my extensions.conf file https://pastebin.com/iEYiSQAr |
| 13:45.42 | flexnsniff | aren't permissions lovely |
| 13:48.11 | flexnsniff | rfr__ - just glancing at the config, it looks alright. In your dahdi.conf, are your contexts set correctly? |
| 13:48.47 | flexnsniff | Are you seeing any feedback from DAHDI inside of the asterisk prompt? |
| 13:48.57 | flexnsniff | and lastly, what is the output of 'dahdi show channels' |
| 13:50.19 | flexnsniff | I had that problem - POTS/loop start signaling was working fine (its done by the card) |
| 13:50.39 | flexnsniff | but since the channel had not been loaded into asterisk, dialed numbers had no dialplan to reach |
| 13:51.10 | flexnsniff | the card realizes there's no upstream signaling from asterisk and throws the phone a fast busy |
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| 14:09.39 | [TK]D-Fender | rfr__, Dialplan doesn't show us what's happening |
| 14:10.06 | [TK]D-Fender | rfr__, Though on a side note that syntax is massively outdated (1.2 grade) and will fail all over the place |
| 14:14.44 | rfr__ | Flexsniff I will check out my dahdi.conf and post. |
| 14:15.43 | rfr__ | [TK]D-Fender I believe I am experiencing the 'fail all over the place' and suggestions on updating it? |
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| 14:19.15 | rfr__ | flexsniff: Ok, /etc/modprobe.d/dahdi.conf is empty. So that's a problem. Any idea what that file was called in the hold zaptel configurations, so I can go look at it for reference? |
| 14:28.08 | [TK]D-Fender | rfr__, for dialplan, "|" isn't a delimiter since 1.4 it's only ",". Voicemail doen't put letter before the box number (read all your app's instructions to verify changes) |
| 14:28.28 | TrickkyTyper | Hey guys what else do i need in ordre to get asterisks making calls |
| 14:28.29 | TrickkyTyper | for example |
| 14:28.45 | [TK]D-Fender | just prove your dahdi channels are loaded at all from CLI |
| 14:28.47 | TrickkyTyper | like I download it, install and configure it, do i need to pay a gateway? How do i actually obtain a static phone number? |
| 14:28.58 | [TK]D-Fender | ~itsp |
| 14:28.58 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:29.00 | [TK]D-Fender | ^^^ |
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| 14:29.03 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:29.11 | [TK]D-Fender | or use some hardware line you're already paying for |
| 14:29.13 | TrickkyTyper | okay so with Asterisks ill still need an ITSP |
| 14:29.20 | TrickkyTyper | Correct? |
| 14:29.45 | TrickkyTyper | That is what im trying to figure out before I jump in. |
| 14:29.45 | [TK]D-Fender | It doesn't provide service out of thin air. * is a toolkit for processing calls |
| 14:29.51 | TrickkyTyper | IO understand that |
| 14:30.05 | [TK]D-Fender | just like buying an answering machine for a POTS line doesn't give you POTS service |
| 14:30.29 | TrickkyTyper | Okay, so now i get it i still need to get an ITSP before i install astersisks and my SIP |
| 14:30.30 | TrickkyTyper | okay |
| 14:30.33 | TrickkyTyper | ty |
| 14:30.37 | TrickkyTyper | i know that was a stupid question |
| 14:30.39 | TrickkyTyper | i just wanted to verify |
| 14:30.47 | [TK]D-Fender | * can talk to a variety of technologies. It's up to you to chose one and configure it |
| 14:30.57 | [TK]D-Fender | Not tha bad, don't worry |
| 14:31.17 | TrickkyTyper | Sorry last question which ITSP do you recommend for a very small company |
| 14:31.18 | TrickkyTyper | cheap |
| 14:31.27 | [TK]D-Fender | depends where you're located |
| 14:31.30 | TrickkyTyper | canada |
| 14:31.31 | TrickkyTyper | personally |
| 14:31.49 | [TK]D-Fender | Flowroute has CA DID's I believe |
| 14:31.54 | [TK]D-Fender | les.net is pretty decent |
| 14:31.56 | [TK]D-Fender | voip.ms |
| 14:32.02 | TrickkyTyper | okay thank you. |
| 14:32.19 | TrickkyTyper | another question |
| 14:32.44 | TrickkyTyper | Cisco SPA122, ATA with Router - Affordable and Feature-Rich Voice over IP (VoIP) would support asterisks right? |
| 14:32.54 | [TK]D-Fender | they both talk SIP |
| 14:32.55 | rfr__ | [TK]D-Fender: Thank you. Where can I find app's instructions? |
| 14:33.07 | [TK]D-Fender | So it will interact as normally as anything else |
| 14:33.26 | [TK]D-Fender | rfr__, "core show application APPNAME" |
| 14:33.37 | [TK]D-Fender | rfr__, and read the official WIKI |
| 14:34.00 | rfr__ | [TK]D-Fender: So just to be clear replace '|' with ',' as a delimiter? |
| 14:34.18 | [TK]D-Fender | everywhere you have it |
| 14:34.35 | rfr__ | Thank you. |
| 14:34.46 | [TK]D-Fender | exten => s,102,Voicemail(b${ARG1}) |
| 14:35.07 | [TK]D-Fender | the concept of priority jumping (+100) is also long dead |
| 14:35.16 | [TK]D-Fender | +101 |
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| 14:39.15 | rfr__ | [TK]D-Fender: how should I edit 'exten => s,102,Voicemail(b${ARG1})' ? |
| 14:40.11 | [TK]D-Fender | Priority jumping is gone. You need to read how the apps that used to jump work again |
| 14:40.21 | [TK]D-Fender | And go read voicemail's instructions |
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| 14:45.06 | rfr__ | Anyone know where I can hire someone to update my dialplan? |
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| 15:01.36 | jeffspeff | I've been reading up on the AGI and found pyst2. What can I do in the dialplan that can't be done in a python AGI? It seems to me that you could get a lot more versatility and functionality by handling all calls in a python based AGI script. Am I missing something? |
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| 15:03.38 | [TK]D-Fender | Not sure how apps that include calling other dialplan operated from there. |
| 15:03.43 | [TK]D-Fender | like Dial w/ macros, etc |
| 15:04.25 | [TK]D-Fender | AGI is a larger load on your system so I would normally advise using it for only the things that require it |
| 15:12.42 | igcewieling | jeffspeff: AGIs are so much more powerful that simple dialplan it can be hard to understand |
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| 15:24.26 | jeffspeff | how much overhead does AGI add? |
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| 15:29.13 | igcewieling | jeffspeff: Spawing a process like Asterisk would do when you start an AGI isn't something you want to for every line of dialplan, but running 1 or 2 agis should be fine. I run one AGI at the start of every call and one AGI at the end of every call. |
| 15:32.19 | TrickkyTyper | hey guys anyone know an ITSP that allows free outgoing calls, or unlimited outgoing calls, and doesnt go by "MInutes" |
| 15:32.38 | TrickkyTyper | IO seem to keep finding some that say like 500 minutes for 6$ a month but im thinking i just would rather pay more and have unlimited calls |
| 15:35.01 | [TK]D-Fender | Many providers have plans like that |
| 15:35.15 | [TK]D-Fender | how much time do you actually expect to be on calls? |
| 15:35.26 | [TK]D-Fender | Note: ALL of those "unlimited" plans ... HAVE limits |
| 15:35.32 | aoeui | TrickkyTyper: https://www.localphone.com/ has some cheap outgoing subscription plans |
| 15:36.01 | [TK]D-Fender | Starting with only supporting 1 or 2 simultaneous calls max then you buy the same product again if you expect multiple people to be on calls |
| 15:36.12 | [TK]D-Fender | at which point they ALL have to be on the phone a lot to hope to be worth it |
| 15:36.55 | TrickkyTyper | thank you so much aoeui |
| 15:37.16 | TrickkyTyper | it will just be basically me right now making calls |
| 15:38.29 | igcewieling | TrickkyTyper: for most ITSPs "unlimited" is not unlimited. IIRC my ITSP limits unlimited accounts to 5,000 mins/month or something like that. |
| 15:38.37 | [TK]D-Fender | Since you can LD around $0.01/min if you look around, go count how many minutes you actually expect to be using... |
| 15:52.41 | jeffspeff | TrickkyTyper, I'm not sure where you are but my personal favorite is https://www.flowroute.com/pricing/ pretty cheap, good quality of service and good support. |
| 15:57.54 | jeffspeff | is there a way to noload all modules and then only load the ones you want? |
| 15:58.22 | jeffspeff | let me clarify, no load all modules without specifying a noload statement for each module |
| 15:58.33 | file | You can turn autoload off |
| 15:58.55 | jeffspeff | should have seen that... thanks file |
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| 17:58.54 | bigbadpig | I had a google voice number set up for years on asterisk. Suddenly I can't hear anyone through it. I can call out and call in and others can hear me but I can't hear them. |
| 17:59.00 | bigbadpig | Have any changes been made? |
| 17:59.12 | file | Google Voice is transitioning to a new SIP based system |
| 17:59.23 | file | the old thing seems to be breaking as a result, and is likely to disappear eventually |
| 18:06.17 | kfife | Wow. Hopefully I can use Asterisk as a client for my GV number using chan_sip |
| 18:09.10 | file | there is someone adding support to it in chan_pjsip |
| 18:11.43 | sibiria | anyone around who might be familiar with changes to asterisk and/or chan_sip between 1.8 and 13, resulting in CDR duration and billsec ending up the same? (i know, big leap between 1.8 and 13) |
| 18:12.18 | sibiria | system is set-up as before, but i can't find anything apparent that could cause this |
| 18:13.40 | bigbadpig | fife thanks a lot. Do you have any links on this? I found this: http://www.dslreports.com/forum/r31939325-Google-Voice-XMPP-support-will-go-away-in-June~start=30 |
| 18:13.57 | file | nope, just DSLReports stuff |
| 18:14.11 | file | Google is Google. It was never a publicly defined interface or service. |
| 18:15.20 | bigbadpig | any links on the chan_pjsip stuff...maybe it's in that link I pasted. I'm just curious about the reverse engineering. Smart folks. |
| 18:15.49 | file | they're on DSLReports, I don't have them handy |
| 18:16.09 | bigbadpig | thanks |
| 18:20.49 | drmessano | I'd do anything to keep from having to pay for phone service |
| 18:20.58 | drmessano | I'll even donate $250 to the project |
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| 18:52.48 | Fwiend_Biggus | drmessano : I don't understand the obsession with free phone service. IT's SO cheap these days. $250 can buy you a DID for 30 years. |
| 18:53.23 | drmessano | Uh, I was being facetious |
| 18:53.28 | Fwiend_Biggus | LOL |
| 18:54.04 | drmessano | I figured the "I'll donate $250 to the project" would have been a dead giveaway |
| 18:54.16 | drmessano | Though the mentality does exist |
| 18:54.31 | Fwiend_Biggus | drmessano : preach it. |
| 18:55.32 | drmessano | The NerdVittleites are foaming at the mouth for this GVSIP shit and it reminds me why I still lurk in telephony forums |
| 18:56.05 | drmessano | GUYS, GUYS... WE DONT HAVE TO GIVE UP GOOGLE VOICE YET. WE MAY BE ABLE TO GET ANOTHER 6 OR 7 MONTHS OUT OF THIS |
| 18:56.16 | drmessano | TAKE MY MONEY |
| 18:56.25 | Fwiend_Biggus | The only reason I want GV on * is because I'm a FI subscriber, and want integration. |
| 18:57.09 | drmessano | I reupped my Flowroute account the other day with $35.. I did it from my tablet, and didnt realize I had pushed the "button" twice because the page didn't redirect |
| 18:57.16 | drmessano | So I ended up paying $105 |
| 18:57.21 | drmessano | and I thought about it |
| 18:57.56 | drmessano | That's like 5 years of service for me |
| 18:58.11 | drmessano | I never use my home phone.. So I did ask for a refund for the additional clicks |
| 18:58.28 | Fwiend_Biggus | Without GV on ast, I can't get a media-path-optimized mobile phone multi-ring to ring to this baby: https://twitter.com/karlfife/status/1013594950175477760 |
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| 18:58.48 | drmessano | Ha nice |
| 18:59.25 | Fwiend_Biggus | The DTMF chip non-invasively counting off the pulses is a nice touch too. |
| 18:59.41 | Fwiend_Biggus | Allows me to press 1 for more options |
| 19:00.20 | drmessano | I do get the whole GV integration. What I don't get is the frothing over the free calling and the urgency to get this back working "or I can't make calls" |
| 19:01.04 | file | having been in the position of being the person who provided such code, the weight and gravity people place upon it is... frustrating at times |
| 19:01.46 | drmessano | file: In the words of Egon Spengler, I blame myself |
| 19:02.02 | drmessano | I am lowkey responsible for it getting included in 1.8 in the first place |
| 19:02.27 | file | frustrating is the wrong word |
| 19:02.28 | file | discouraging |
| 19:02.31 | Samot | Oh, Chan_Motif is your fault? |
| 19:02.32 | file | that's better |
| 19:02.33 | Samot | Figures. |
| 19:02.46 | Samot | drmessano: You're why we can't have nice things. |
| 19:03.19 | Samot | Jerk. |
| 19:03.46 | drmessano | I need to find that conversation. I logged it. Where I told Russell "Hey man, I know 1.8 is code freeze right now, but there's this patch to make this new Google Voice thing work with XMPP/Jingle in Asterisk. This would be REALLY COOL if you could get it into 1.8. PLEAAAASEE.. here's the link to the bug in the tracker:" |
| 19:03.56 | drmessano | More or less he went and merged it after I asked him |
| 19:04.14 | drmessano | It wasn't my patch or anything, I just wanted it |
| 19:04.16 | drmessano | and he caved |
| 19:04.22 | drmessano | and I blame myself |
| 19:05.20 | drmessano | So, file, I am sorry |
| 19:05.29 | drmessano | I guess it would have made it in anyway |
| 19:05.32 | drmessano | But, you know |
| 19:06.43 | drmessano | With all this drama lately, it was ironic that I made such a big deal about it back then lol |
| 19:06.56 | jpsharp | Everyone grows up sometime :) |
| 19:08.13 | drmessano | Well, I was 33 at the time.. so yeah |
| 19:08.33 | drmessano | I had to look back and see when 1.8.0 was released.. wow.. |
| 19:16.17 | TrickkyTyper | hey guys if i downloaded something like ring on my computer like a SIP softphone application what ITSP or Voip provider would i use to give me a phone number |
| 19:16.24 | TrickkyTyper | so i can personally make calls off my computer |
| 19:16.57 | drmessano | ~itsplist |
| 19:17.14 | drmessano | ~itsplist-us |
| 19:17.14 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 19:18.02 | TrickkyTyper | so it is an ITSP |
| 19:18.06 | TrickkyTyper | im looking for or SIP provider. |
| 19:18.40 | drmessano | Ok, and there is a list of some |
| 19:19.29 | TrickkyTyper | ok |
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| 19:46.02 | jeffspeff | anyone know why flowroute is listed on there as least respected? |
| 19:46.23 | [TK]D-Fender | That's not erally the intended message |
| 19:46.31 | [TK]D-Fender | really |
| 19:46.41 | jeffspeff | ah, ok |
| 19:46.55 | [TK]D-Fender | Not sure when Junction made it on there though |
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| 21:10.35 | drmessano | I'm not sure I would even consider the last one on the list "least respected" anyway |
| 21:10.46 | drmessano | Considering it's a list of popular ones |
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| 22:53.59 | salviadud | Wanna see my cat? |
| 22:54.06 | salviadud | ________ââââââââââââââââââââ______ |
| 22:54.06 | salviadud | _______ââââââââââââââââââââââ_____ |
| 22:54.06 | salviadud | _______ââââââââââââââââââââââââââ_ |
| 22:54.09 | salviadud | __ââ___ââââââââââââââââââââââââââ_ |
| 22:54.12 | salviadud | âââââââââââââââââââââââââââââââââ_ |
| 22:54.15 | salviadud | _ââââââââââââââââââââââââââââââââ_ |
| 22:54.18 | salviadud | _____âââââââââââââââââââââââââââââ |
| 22:54.21 | salviadud | ______âââââââââââââââââââââââââââ_ |
| 22:54.24 | salviadud | ____âââââââââââââââââââââââââââ___ |
| 22:54.27 | salviadud | ____ââââ_ââââ_______ââââ_ââââ_____ |
| 23:01.50 | Samot | Probably should have waited for a "yes" before flooding the channel. |
| 23:16.58 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-rfxvixujdgifzgxy) |
| 23:27.11 | *** join/#asterisk mhache (~mhache_@198.164.250.208) |
| 23:28.54 | *** join/#asterisk robmal (r@wporzo.pl) |