| 00:07.27 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 00:13.28 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 00:19.01 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:19.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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 |
| 00:33.24 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 01:11.28 | snadge | curious.. i have an asterisk 11 based pbx, which if it receives a call from one of our core servers, it has a remote party id header in it |
| 01:11.53 | snadge | and when it rings the extension, it changes the contact header, to the number in the rpid.. which is what it wanted |
| 01:12.22 | snadge | because when you try to call back the missed call on the device, it calls back the remote party... instead of the trunk username (which obviously doesn't work) |
| 01:13.03 | snadge | our other core server sends an inbound call without the remote party id header, to the pbx, and when the call is forwarded to the extension.. the contact field contains the trunk username |
| 01:14.20 | snadge | the difference between the core servers is.. the working one is asterisk 1.6, the other is asterisk 11.. probably a few other config differences as well |
| 01:15.41 | snadge | .. is there a setting which will add rpid header to incoming calls? |
| 01:18.51 | snadge | hmm.. its configured sendrpid=pai .. and yet there is no pai |
| 01:22.31 | snadge | ah fuck it .. i'll figure it out myself.. thats what i get paid for right? :P |
| 01:23.14 | snadge | this entire asterisk infrastructure is eventually going to be binned for freeswitch anyway ;) |
| 01:28.19 | Samot | Ok |
| 01:28.42 | Samot | Pai is a callerid format |
| 01:29.08 | Samot | Sendrpid=pai will format the callerid in PAI. |
| 01:29.58 | Samot | RPID and PAi |
| 01:30.17 | Samot | Are still things on FreeSwitch |
| 01:31.04 | snadge | i figured out the complicating factor.. we use a mysql asterisk realtime db |
| 01:31.35 | snadge | and of course the sipconf table has completely different columns between version 1.6 and 11 |
| 01:32.23 | snadge | so in the asterisk 11 table.. the sendrpid column (which doesn't even exist in the asterisk 1.6 version), is set to no for the particular peer |
| 01:32.44 | Samot | Which will makeit use FROM |
| 01:32.58 | snadge | problem is.. from looks like this |
| 01:33.27 | snadge | "phone_number" <username@host>;etc |
| 01:33.37 | Samot | Right. |
| 01:33.48 | snadge | so of course the phone picks the username as the calling party.. because its stupid |
| 01:33.50 | Samot | That looks like a proper FROM tag. |
| 01:34.15 | snadge | so the workaround seems to be.. if the contact header contains the phone number, instead of the username |
| 01:34.23 | snadge | it would appear to use that |
| 01:34.39 | snadge | and if the incoming call to the pbx contains the remote party id header.. then it will set the contact header to that, instead of the username |
| 01:34.54 | snadge | so.. the easiest solution would appear to be, enable sendrpid=yes for that peer |
| 01:35.07 | Samot | That would send Remote Party ID |
| 01:35.23 | Samot | Or you can set it to PAI and it will send it in that format. |
| 01:35.39 | snadge | hmm.. yes.. that will work too.. i'll try it |
| 01:36.02 | Samot | These issues will still exist in FS. |
| 01:51.46 | snadge | hey not my decision |
| 01:52.11 | snadge | the new overlords use freeswitch, so it will make sense for them to move core call switching/routing over to it |
| 01:52.18 | snadge | pbxs.. i doubt that will ever happen |
| 01:52.51 | Samot | I wouldn't do switching/routing on Asterisk myself. |
| 02:04.45 | Maliuta_ | snadge: is that Krang or Kodos? |
| 02:04.50 | Maliuta_ | (the overlord) |
| 02:15.41 | snadge | does it matter? :P |
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| 02:18.10 | Maliuta_ | snadge: they are both likely to enslave you, so I guess not :P |
| 02:26.21 | snadge | as long as they pay me.. im really not that passionate about voip :P |
| 02:27.24 | snadge | turns out some monkey put a set callerid in the hunt |
| 02:27.50 | snadge | delete that.. boom.. works perfectly.. after reverse engineering the entire core network, pbx, and recreating the entire setup as a test case |
| 02:27.57 | snadge | :P |
| 02:28.32 | snadge | yay its friday :D |
| 02:32.05 | Samot | You just didn't follow the call through? |
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| 03:03.12 | wolfmitchell | How do I disable all the "SIP read from" messages showing in asterisk -r spamming my console? I tried 'sip set debug off' but I'm still getting spammed with them |
| 03:03.42 | Samot | Show an example. |
| 03:03.59 | wolfmitchell | 1sec, lemme get rid of stuff like IPs and names from it |
| 03:04.47 | wolfmitchell | Samot, https://screenshits.nofla.me/2017-03-16_23-04-40.txt |
| 03:05.33 | Samot | Are you using Chan_SIP? |
| 03:05.37 | wolfmitchell | ye |
| 03:05.46 | Samot | That notify should only be there if the SIP debug is on. |
| 03:06.09 | wolfmitchell | is there anything in sip.conf that would do it? |
| 03:06.17 | wolfmitchell | ... oh wait, i just noticed... sipdebug=yes in it |
| 03:06.18 | wolfmitchell | lmao |
| 03:06.21 | wolfmitchell | that would do it |
| 03:06.24 | Samot | Well yeah, that would do it. |
| 03:06.38 | wolfmitchell | note to self: read the config files before going to irc |
| 03:07.04 | drmessano | Yeah, dumbass |
| 03:07.06 | drmessano | Im kidding |
| 03:07.16 | wolfmitchell | yay now i can actually read stuff in my console without it being pushed off screen in a second :P |
| 03:07.18 | wolfmitchell | drmessano, lol |
| 03:07.25 | wolfmitchell | oh look my childhood nickname! /s |
| 03:07.47 | drmessano | I spent 12 years thinking that was my name |
| 03:07.54 | wolfmitchell | hahaha |
| 03:07.54 | drmessano | So really, the joke is on me |
| 03:08.04 | drmessano | Screw you, Dad |
| 03:08.04 | wolfmitchell | anyway, i gotta go |
| 03:08.08 | drmessano | Peace out |
| 03:12.14 | wolfmitchell | aand now i'm on my laptop :P |
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| 05:24.21 | ruben23 | hi there guys wanted to ask regarding Dial status on asterisk, what specifically CHANNELUNAVAILABLE Means..? when we dial and get this STATUS from a VoIP Carrier.. Any idea guys.? Thanks in advance |
| 05:26.38 | Samot | It means there are no available channels. |
| 05:27.45 | Samot | Does the carrier limit your voice channels? |
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| 05:41.22 | ruben23 | Samot: im openned with 1000 channels |
| 05:42.34 | Samot | asterisk -rvvvvvvvvvv |
| 05:42.36 | Samot | sip set debug on |
| 05:42.40 | Samot | show a call that gets the error. |
| 05:42.44 | Samot | ~pb |
| 05:42.45 | infobot | i heard 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. |
| 05:49.25 | ruben23 | ok |
| 05:58.13 | igcewieling | check the value of ${HANGUPCAUSE} and ${HANGUPCAUSE(0,tech)} or maybe ${HANGUPCAUSE(1,tech)} and see if that helps |
| 05:58.50 | igcewieling | DIALSTATUS is not granular enough for troubleshooting. HANGUPCAUSE is better. |
| 05:59.26 | igcewieling | You need to do a packet capture to see what the ITSP is sending back to you. |
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| 06:43.24 | Samot | Well I waited an hour to see a debug of a call. Didn't get one. I'm going to bed. |
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| 08:47.05 | chl_ | hello, can you change the log level of odbc log in /tmp/? |
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| 11:02.17 | sarthor | Hi, I have dynammic IP on my DSL line, I am using dyndns for reaching my pbx, IN my DSL modem setting i have created DMZ and my pfsense firewall IP is pointed as DMZ, I am doing NATing on my pfsens toward my IPPBX machine, my local extensions can connect easily, from Internet side only one softphone Zoiper can connect, non of other soft or hard phone working. HELP. |
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| 11:13.53 | sotoz | hello, I use a stasisapp and ARI and I'm creating two channels which I add to a bridge and then I dial both. The thing is that I only get one new channel on asterisk on Ring state which is the one that dials one of the numbers. The channels that I create/dial are in the Down state. How can this happen? |
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| 14:51.43 | sarthor | Hello, I am using DSL line with dynamic IP, and my pfsense router is in DMZ of my modem, I have ippbx installed, VM ippbx-->pfsense--dsl modem--Internet, I have extension on local lan as well as on Internet side. was just asking what will be better to go with, Configuring stun server on my VM, or just port forwarding stuff, I have face problems in NATing stuff. (I am using dyndns host due to dynamic IP) |
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| 15:28.32 | igcewieling | what version of Asterisk? |
| 15:29.53 | Samot | He's on FreePBX. |
| 15:29.57 | Samot | Just to throw that out there. |
| 15:30.26 | Samot | Plus, PFSense. Horrible for VoIP. |
| 15:30.53 | Samot | <PROTECTED> |
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| 15:36.10 | lvlinux | Samot: how is pfSense horrible for VoIP? |
| 15:36.33 | lvlinux | (after fixing static UDP NAT port mapping) |
| 15:37.55 | Samot | The SIP helpers don't help. |
| 15:38.06 | Samot | The port randomization is wonky. |
| 15:38.41 | lvlinux | SIP helpers are optional to install, and all you have to do is turn of port randomization and it works fine for me at least. |
| 15:38.49 | Samot | People come into #freepbx all the time with NAT problems. Not all people with NAT problems are running pfSense but all pfSense users have NAT issues. |
| 15:40.31 | lvlinux | Well if they run it vanilla then yes I can understand that, but turning off port randomization for UDP is a simple fix. |
| 15:41.33 | Samot | That's why I said do those two things. |
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| 15:56.30 | *** join/#asterisk mrzzzip (~test@pool-96-239-66-20.nycmny.fios.verizon.net) |
| 15:56.39 | mrzzzip | hi, |
| 15:57.44 | mrzzzip | can someone help me troubleshoot x100p box, I cant figure out if incoming calls are coming in or not |
| 15:58.35 | Samot | What's a x100p box? |
| 15:59.19 | newtonr | that is old school https://www.voip-info.org/wiki/view/X100P+clone |
| 15:59.47 | mrzzzip | http://www.x100p.com/products/js200fx.php |
| 16:00.15 | Samot | The most stable Asterisk v1.4 branch with AcceleratedGUI for Asterisk |
| 16:00.19 | Samot | ^^^^ Best part of that page. |
| 16:00.57 | mrzzzip | yeah well its a toy, what can i say |
| 16:01.05 | Samot | "The most stable" |
| 16:01.09 | Samot | Of v1.4 |
| 16:01.17 | Samot | Versus all the other non-stable version of v1.4 |
| 16:01.20 | Samot | That were release. |
| 16:01.25 | Samot | Anyways, we can try |
| 16:01.26 | igcewieling | you won't find many people here willing or able to help with x100p clones. |
| 16:01.30 | Samot | But 1.4 is... |
| 16:02.04 | mrzzzip | ok i i got it up and running and i can make outgoing calls with a Trunk, I just cant incoming calls |
| 16:02.06 | Samot | mrzzzip: do you have cli access to the box? |
| 16:02.15 | *** part/#asterisk igcewieling (~ewieling@162-236-85-155.lightspeed.moblal.sbcglobal.net) |
| 16:02.15 | mrzzzip | gui cli |
| 16:02.21 | Samot | Ugh. |
| 16:02.32 | Samot | And sip debug is not great in 1.4 |
| 16:02.40 | mrzzzip | let me try to see if there another way, like ssh? |
| 16:02.44 | Samot | How do you peer with the provider to get incoming calls. |
| 16:02.46 | Samot | Yes, SSH. |
| 16:04.24 | mrzzzip | no |
| 16:04.30 | Samot | How do you peer with the provider to get incoming calls? |
| 16:04.35 | mrzzzip | trunk |
| 16:04.38 | *** join/#asterisk u0m3 (~u0m3@79.115.167.77) |
| 16:04.40 | Samot | Do you register? |
| 16:04.43 | mrzzzip | yes |
| 16:04.51 | Samot | So you can register without issue? |
| 16:05.12 | mrzzzip | i didnt verify on the provider end, but i can make outgoing calls |
| 16:05.42 | Samot | Outgoing is different. |
| 16:05.46 | Samot | Auth is done per call. |
| 16:05.54 | lvlinux | mrzzzip: registration is what allows the provider to send you incoming calls. |
| 16:05.58 | Samot | ^^^ |
| 16:06.03 | Samot | They need your location. |
| 16:06.06 | Samot | IP:port |
| 16:06.13 | lvlinux | Is res_xmpp able to do SRV lookups?? I have dnsmgr and unbound on v14 |
| 16:06.37 | Samot | What happens when you make an incoming call? |
| 16:08.12 | *** join/#asterisk freebs (~freebs@gateway/vpn/privateinternetaccess/freebs) |
| 16:08.24 | lvlinux | nevermind what i said....didn't have the config file in the right directory.... |
| 16:09.20 | mrzzzip | ok its not registering |
| 16:09.42 | Samot | That would be the issue. |
| 16:10.47 | mrzzzip | i can register with freepbx, how do i compare the 2 |
| 16:11.52 | mrzzzip | ok, got it to register now |
| 16:12.07 | mrzzzip | from what i can see in. sip show registry |
| 16:13.51 | Samot | So FreePBX can register but the x100p can't? |
| 16:14.15 | mrzzzip | i got the x100p to register |
| 16:14.18 | Samot | K |
| 16:14.25 | mrzzzip | but when i call in .. i just get dead silence |
| 16:14.35 | mrzzzip | i used "s" for the incoming string |
| 16:14.44 | mrzzzip | which is a catchall |
| 16:15.13 | mrzzzip | is there a way to see if the call is hitting the box? |
| 16:15.24 | lvlinux | using SSH and the console, yes |
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| 16:15.29 | [TK]D-Fender | mrzzzip> i got the x100p to register <- thats a CARD |
| 16:15.39 | [TK]D-Fender | how the hell does a PCI card "register"? |
| 16:16.03 | lvlinux | lol i think mrzzzip means the "box with the x100p" |
| 16:16.49 | lvlinux | I'm still having an issue with DNS--- does res_xmpp use unbound? it won't do an srv lookup for me. |
| 16:16.53 | mrzzzip | x100p is a box |
| 16:16.54 | mrzzzip | :) |
| 16:17.29 | lvlinux | I'm getting: ERROR[29315]: res_xmpp.c:3636 xmpp_client_reconnect: No DNS available for XMPP connection when trying to connect client 'asterisk-ruhnet.net' |
| 16:18.49 | [TK]D-Fender | S200-FX v2.0 Embedded VoIP PBX Asterisk Appliance |
| 16:18.53 | Samot | I'm not sure about that...for the DNS. |
| 16:18.56 | [TK]D-Fender | That doesn't say "x100p" to me |
| 16:19.29 | mrzzzip | the website is x100p.com |
| 16:19.31 | drmessano | lvlinux: Im pretty sure DNS SRV records for XMPP in Asterisk worked fine |
| 16:19.54 | lvlinux | when I specify the actual fqdn of the specific xmpp server it works, but when i just specify the domain with the srv record that points to the particular server, it complains. |
| 16:20.06 | [TK]D-Fender | mrzzzip, their website isn't doing the registering |
| 16:20.12 | lvlinux | (and i did verify that the srv record was correct) |
| 16:20.22 | drmessano | lvlinux: "does res_xmpp use unbound" <-- No, it uses DNS. Unbound is your choice. res_xmpp doesn't care which DNSd you have |
| 16:20.33 | [TK]D-Fender | When you ask what kind fo computer I have the answer "NewEgg" because that's the site I bought it from means nothing. |
| 16:21.32 | mrzzzip | ok so I need help with the s200-fx box |
| 16:22.56 | lvlinux | drmessano: that's what I thought, thanks. |
| 16:23.04 | Samot | Will XMPP add the srv prefix for lookups? |
| 16:23.31 | Samot | I know you have to turn that on for ChaN_SIP (not sure about PJSIP) |
| 16:23.45 | lvlinux | i didn't see an option for it in the sample xmpp.conf |
| 16:23.50 | Samot | Neither do I. |
| 16:23.56 | lvlinux | that was the first thing I checked |
| 16:24.36 | Samot | SRV lookups have to be done in a specific way. |
| 16:24.46 | Samot | That's why there's NAPTR records. |
| 16:27.22 | mrzzzip | so, im back where i started.. how do i begin to troubleshoot the no incoming calls. just dead silence when i dial the number |
| 16:27.32 | *** join/#asterisk miralin1 (~Thunderbi@194.8.128.48) |
| 16:27.42 | Samot | We would have you SSH into the box and do debugs. |
| 16:27.44 | drmessano | Samot: I believe even the former XMPP module did it |
| 16:27.51 | drmessano | Because it's expected |
| 16:27.56 | Samot | Yeah. |
| 16:28.48 | Samot | lvlinux: So you have something like sub00.domain.tld that has SRV records for like sub01 and sub02? |
| 16:29.31 | drmessano | lvlinux: Is that the actual domain? |
| 16:29.36 | drmessano | The one you pasted? |
| 16:29.48 | drmessano | Because it's failing the lookup test |
| 16:30.38 | Samot | Same here |
| 16:31.44 | lvlinux | Samot, drmessano: ruhnet.net is the domain with the SRV records, raven.ruhnet.net is the actual xmpp server where the SRV records point to. |
| 16:32.09 | lvlinux | asterisk-ruhnet.net is just the section name in the config file |
| 16:32.34 | sarthor | lvlinux: Samot SIP helpers means "SipDroid" on pfsense? removed. ports already forwareded |
| 16:33.20 | Samot | I can't resolve ruhnet.net but I can resolve raven.ruhnet.net with a simple A record lookup. |
| 16:33.32 | lvlinux | when i put serverhost=raven.ruhnet.net in xmpp.conf and reload it, it connects immediately, but when i just put ruhnet.net, i get one error every 4 seconds or so complaining about "No DNS available for XMPP connection..." |
| 16:33.49 | drmessano | lvlinux: Ok, that passes. |
| 16:34.08 | file | res_xmpp doesn't do DNS, it's up to the underlying client library - iksemel |
| 16:34.28 | lvlinux | aha! |
| 16:34.48 | lvlinux | does iksemel have any settings somewhere? |
| 16:35.12 | file | the only settings that you can set would be done from the XMPP configuration file |
| 16:35.35 | lvlinux | k |
| 16:36.20 | sarthor | says this Request 'REGISTER' from '<sip:102@vxxx.xxx.com>' failed for '95.187.129.120:1576' (callid: dTi45bbiRyEswwMkQT7mgw..) - No matching endpoint found |
| 16:36.53 | sarthor | dialing from DLINK IP Phone |
| 16:37.08 | sarthor | Connecting* |
| 16:37.51 | sarthor | and Ekiga says "Could no register (Method not allowed)" |
| 16:37.55 | Samot | lvlinux: So your SRV record looks like _xmpp-client._tcp.raven.ruhnet.net? |
| 16:38.17 | Samot | lvlinux: And _xmpp-server._tcp.raven.ruhnet.net? |
| 16:38.20 | drmessano | Samot: That's not correct |
| 16:38.28 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d1f:fc17:9f4:53c2:5079:b79a) |
| 16:38.36 | Samot | What should it be then? |
| 16:38.50 | drmessano | The host is raven.. the record points to the domain, not the host |
| 16:38.51 | Samot | rfc6120 <-- something change? |
| 16:38.54 | drmessano | What he has is correct |
| 16:39.06 | Samot | I can't get a result. |
| 16:39.11 | drmessano | _xmpp-client._tcp.ruhnet.net |
| 16:39.51 | Samot | Got it. |
| 16:39.55 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 16:40.10 | Samot | Problem is, iksemel has to slap that on. |
| 16:40.18 | Samot | _xmpp-client._tcp. |
| 16:40.40 | Samot | Otherwise it's just going to look up ruhnet.net |
| 16:40.51 | drmessano | Of course |
| 16:41.12 | lvlinux | yes |
| 16:42.01 | lvlinux | surely iksemel supports that??? |
| 16:42.18 | drmessano | lvlinux: It's been a while, but I just checked.. I am using the FQDN |
| 16:42.26 | drmessano | So probably not |
| 16:42.45 | *** join/#asterisk u0m3 (~u0m3@79.115.167.77) |
| 16:42.53 | drmessano | If DNS SRV worked, I wouldn't have used the FQDN |
| 16:43.23 | lvlinux | eeek |
| 16:43.44 | Samot | You can try a NAPTR record. |
| 16:43.58 | Samot | It may try that first before a ip4 A record check. |
| 16:44.13 | drmessano | IIRC this particularly sucked because I had multiple XMPP servers and I wanted it to follow which primary > secondary on failure |
| 16:45.14 | lvlinux | i only have the one now, but was planning to have backups in future. |
| 16:45.36 | lvlinux | Yes it appears that iksemel doesn't support it---google shows problems with that elsewhere too. |
| 16:45.36 | drmessano | Ultimately I just created 2 XMPP clients and connected them to server 1 and server 2 |
| 16:45.44 | lvlinux | yuck |
| 16:45.49 | drmessano | Its all coming back to me now |
| 16:47.28 | drmessano | So basically I had call notifications being sent by both clients |
| 16:47.35 | drmessano | and they would just fail on the backup |
| 16:47.52 | drmessano | Because all my other clients followed SRV priority |
| 16:48.09 | drmessano | Turn off logging on the backup lol |
| 16:49.16 | lvlinux | lol |
| 16:49.41 | lvlinux | oh well, at least i know it's not a misconfig issue now. so I'll just put in the actual host. |
| 16:50.42 | drmessano | Yeah the DNS record looks fine |
| 16:50.56 | drmessano | SO you shouldnt have any problems from other clients |
| 16:51.19 | lvlinux | no i haven't so far. even with federation |
| 16:51.39 | drmessano | What are you using for the XMPP server? |
| 16:52.02 | lvlinux | now i need to figure out the gtalk one---pretty sure it is a "security" issue with them as it spits out tons of errors for that... |
| 16:52.09 | lvlinux | using Prosody for th eserver |
| 16:52.18 | drmessano | Cool.. same here |
| 16:52.22 | Samot | I thought Prosody did SRV? |
| 16:52.28 | lvlinux | it does |
| 16:52.31 | drmessano | It does |
| 16:54.11 | lvlinux | with the gtalk client i keep getting TONS of res_xmpp.c:3735 xmpp_client_receive: Parsing failure: Invalid XML |
| 16:54.17 | lvlinux | and then res_xmpp.c:3802 xmpp_client_thread: JABBER: socket read error |
| 16:54.38 | lvlinux | and I'm thinking it's something to do with their auth... |
| 16:54.51 | drmessano | Are you using an example from the web? |
| 16:54.56 | drmessano | Its pretty well documented |
| 16:56.09 | lvlinux | from the Asterisk book |
| 16:56.37 | drmessano | Which edition? |
| 16:57.37 | lvlinux | 4th |
| 16:58.36 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 16:59.03 | drmessano | Just make sure it refers to chan_motif and res_xmpp |
| 16:59.06 | drmessano | not the old modules |
| 16:59.36 | lvlinux | yes of course. the example there is the same as in the book (except for the priority) |
| 17:00.10 | lvlinux | i don't care about motif, just the xmpp part |
| 17:00.24 | lvlinux | or rather to be correct, the IM part |
| 17:00.58 | Samot | Kinda funny I've been working on SRV records during all this. |
| 17:01.25 | lvlinux | hehe |
| 17:02.01 | Samot | Bringing a pair of DNS servers up to have all the devices use them. |
| 17:03.42 | Samot | Mainly because most DNS services dont let you do NAPTR records. |
| 17:04.03 | Samot | Because they can get really complicated. |
| 17:05.04 | drmessano | Im trying to talk myself out of running a DNS server again |
| 17:05.15 | Samot | Well it only does the SIP domains. |
| 17:05.36 | Samot | They aren't meant to be recursive. |
| 17:05.36 | drmessano | I really want my own DNS server for Dynamic DNS |
| 17:05.42 | lvlinux | I use dns.he.net for most stuff |
| 17:05.50 | drmessano | I cant use HE |
| 17:05.52 | drmessano | lol |
| 17:05.56 | lvlinux | why? |
| 17:06.00 | drmessano | Funny story |
| 17:06.12 | lvlinux | oh? |
| 17:06.12 | *** join/#asterisk vstemen (~vince@ip72-202-187-29.ok.ok.cox.net) |
| 17:06.18 | drmessano | So you know how google has their 1e100 domain or whatever it is? |
| 17:06.23 | lvlinux | yeah |
| 17:06.43 | drmessano | I have a numeric domain I use for backend stuff.. i've had it for 15 years |
| 17:07.04 | drmessano | The guys at HE were VERY nice about it.. but they can't allow me to use it |
| 17:07.19 | lvlinux | why not? |
| 17:07.34 | drmessano | Because numeric domains are highly abused |
| 17:07.49 | lvlinux | i didn't know that |
| 17:08.10 | lvlinux | why would a numeric domain be any more abused than a non-numeric?? |
| 17:08.13 | drmessano | I've had lots of offers for it |
| 17:08.39 | drmessano | Because numeric domains are kind of ummm |
| 17:08.42 | drmessano | ambigious |
| 17:08.50 | drmessano | As in.. |
| 17:09.04 | lvlinux | oh i get it |
| 17:09.08 | drmessano | 1e100 looks like some backend something domain that's none of my business |
| 17:09.24 | drmessano | so 6254.com could be some legit Yahoo domain for all I know |
| 17:09.42 | drmessano | backend.routing.6254.com |
| 17:10.07 | drmessano | Looks like "Internet mumbo jumbo" to the uneducated crowd |
| 17:10.13 | lvlinux | yeah |
| 17:10.33 | drmessano | Granted, same people that would click on yah00.com links |
| 17:10.36 | drmessano | But still |
| 17:10.51 | Samot | Never really was a fan of numeric domains. |
| 17:10.59 | Samot | Never offered it. |
| 17:11.06 | drmessano | Some guy wanted to give me $4500 bucks for it |
| 17:11.13 | drmessano | Which was likely a scam |
| 17:11.20 | lvlinux | whoa that's a decent sum, but yeah probably a scam |
| 17:11.22 | drmessano | But I consider the offer more interesting |
| 17:11.33 | drmessano | They would get the domain and I wouldnt see a penny |
| 17:11.45 | drmessano | But them seeking it out is interesting. I get a lot of offers for it |
| 17:12.55 | Samot | I never liked the "domain traders" |
| 17:13.06 | Samot | Its like day trading... |
| 17:13.23 | Samot | Just a bunch of money shuffling.. |
| 17:13.42 | lvlinux | the legit ones are---the others are, well, not so pleasant even as that. |
| 17:15.24 | lvlinux | but anyway i love he.net 50 domains free, dynamic dns support, web interface config (not that I mind editing txt files tho), a great network, and they support SRV/NAPTR/TXT etc. etc. |
| 17:15.43 | lvlinux | never had a single issue with it. |
| 17:17.26 | drmessano | Oh I dont doubt |
| 17:17.28 | drmessano | Oh I dont doubt it |
| 17:17.30 | drmessano | I love he |
| 17:17.52 | drmessano | I wish I could talk them into allowing me to use my domain |
| 17:18.03 | Samot | This will be the first time in like 6 years I've had my own DNS servers. |
| 17:18.36 | *** join/#asterisk vince1 (~vince@ip72-202-187-29.ok.ok.cox.net) |
| 17:19.52 | Samot | But I mean, it's like $25/month for me to have them and that cost is easily absorbed by clients. |
| 17:20.26 | Samot | So yeah, now I have to be responsible for them but then it's on *me*. |
| 17:20.56 | Samot | And not me relaying "it's being worked on." |
| 17:22.12 | drmessano | I should hit up He again |
| 17:22.33 | Samot | Maybe they're more lax. |
| 17:22.45 | *** join/#asterisk libardi (~libardi@187.64.228.189) |
| 17:23.37 | lvlinux | yeah give it a shot---imho it sounds kindof overprotective in a silly way to not allow numeric domains when there are so many non-numeric that can be even more easily abused. |
| 17:24.19 | drmessano | Im trying to go back a few years and remember the process |
| 17:24.37 | lvlinux | Samot why can't you use HE? |
| 17:24.39 | drmessano | it was probably 2014 when I moved from my last DNS hosting |
| 17:25.38 | *** join/#asterisk igcewieling (~ewieling@162-236-85-155.lightspeed.moblal.sbcglobal.net) |
| 17:26.23 | drmessano | I really hate users sometimes |
| 17:26.40 | drmessano | We're getting ready to replace and old scheduling system with a new one |
| 17:26.51 | drmessano | The old one is Sharepoint based and limited as hell |
| 17:26.56 | drmessano | So they had an issue come up today |
| 17:27.15 | drmessano | "Any we can give __ access to the ___ so he can check the schedule" |
| 17:27.34 | drmessano | No, but in a few weeks this won't matter.. we're moving to ____ |
| 17:27.49 | lvlinux | lol |
| 17:27.57 | drmessano | ..and ___ will just send an email when she puts in a new event |
| 17:28.23 | drmessano | So ___ replies "IM NOT GONNA SIT HERE AND SEND AN EMAIL BLAH BLAH BLAH BLAH" |
| 17:28.37 | drmessano | "No, when you add an event, there's a button you push that fires an email to them" |
| 17:28.57 | drmessano | "Oh.. yeah, ok.. that.. I was gonna say... im not sending them an email too" |
| 17:29.14 | drmessano | ... Welll WTF do you consider that BUTTON? |
| 17:30.08 | drmessano | Can we get some more outrage in the house, please? |
| 17:35.11 | lvlinux | man! i'm still getting all these Parsing failure: Invalid XML and JABBER: socket read error messages... |
| 17:36.01 | lvlinux | wait---no, i'm not! |
| 17:36.04 | lvlinux | yay! |
| 17:39.31 | *** join/#asterisk tuxian_ (~tuxian@igilmour.plus.com) |
| 17:40.42 | lvlinux | Evidently turning on the "allow less secure apps" in the google account takes a while to take effect. |
| 17:48.13 | mrzzzip | what file is the trunk inof stored in |
| 17:48.17 | mrzzzip | info |
| 17:53.31 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:53.32 | igcewieling | nowhere, but sip peer info is in /etc/asterisk/sip.conf |
| 17:53.58 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 17:56.47 | lvlinux | barf! |
| 17:56.55 | lvlinux | now the errors are back after a restart |
| 17:57.31 | lvlinux | won't authenticate for some reason. But it showed connected before. |
| 17:58.06 | lvlinux | mrzzzip: "sip peer" is the proper name for a "trunk" which doesn't really have specific meaning. :) |
| 17:58.41 | igcewieling | lvlinux: I blame FreePBX for that. 8-| |
| 17:58.46 | lvlinux | yup |
| 17:59.04 | lvlinux | Now reloading res_xmpp crashes * |
| 17:59.19 | lvlinux | or locks it up rather. |
| 18:03.07 | lvlinux | xmpp show connections for the google account keeps cycling between "waiting to request TLS" "requesting TLS" "waiting to authenticate" "authenticating" "disconnected" |
| 18:06.38 | lvlinux | But it was connected a few minutes ago before I restarted. frustrating... |
| 18:08.00 | drmessano | igcewieling: Im sure it had nothing to do with vendors referring to it as a "SIP Trunk" and the software trying to simplify things |
| 18:08.16 | mrzzzip | does it matter if the trunk info is in sip.conf vs users.conf |
| 18:09.08 | [TK]D-Fender | yes |
| 18:09.12 | [TK]D-Fender | ~users.conf |
| 18:09.12 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 18:11.52 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 18:13.18 | Samot | Lol. |
| 18:13.23 | Samot | I hadn't seen that one. |
| 18:13.55 | lvlinux | lol |
| 18:14.04 | drmessano | Im gonna start symlinking siptrunk.conf to sipt.conf just to piss people off |
| 18:14.12 | drmessano | sip.conf |
| 18:14.21 | Samot | Hah. |
| 18:15.12 | Samot | Trunk is one of them Telecom terms that doesn't belong in VoIP. |
| 18:15.36 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 18:16.06 | drmessano | https://www.twilio.com/blog/2015/09/secure-trunks.html |
| 18:16.15 | drmessano | Samot: Go fuckyaself, basically |
| 18:16.23 | drmessano | Twilio says it's a "TRUNK" |
| 18:16.35 | drmessano | So basically, Websters |
| 18:17.09 | Samot | Well if Twilio says it. |
| 18:17.37 | drmessano | YESTHEYDO |
| 18:18.51 | drmessano | But it's FreePBX's fault |
| 18:19.00 | Samot | Then I must concede. |
| 18:19.14 | drmessano | Not the 595,000 Google results |
| 18:19.21 | drmessano | Pointing to providers |
| 18:20.23 | Samot | Actually... |
| 18:20.35 | Samot | ; When setting up trunks, make sure there's no risk that any From: username (caller ID) will match any of your device names, because then Asterisk might match the wrong device. |
| 18:20.39 | Samot | ^^^^ I blame Asterisk. |
| 18:23.25 | lvlinux | anybody use (and like) Twilio SIP, uh channels? |
| 18:23.27 | lvlinux | lol |
| 18:23.54 | lvlinux | i'm not really interested in the programming stuff, just the SIP |
| 18:24.13 | drmessano | I wouldnt use it for SIP unless I was high volume |
| 18:24.25 | drmessano | Flowroute is cheaper.. so is VoIP.ms, I think |
| 18:24.45 | drmessano | But even then, both of those have rates that scale |
| 18:24.58 | drmessano | The value in Twilio is their API |
| 18:25.30 | lvlinux | i thought twilio SIP was something like $.0045/min out and $.007 in? |
| 18:26.09 | drmessano | .015 out |
| 18:26.18 | drmessano | .0075 in |
| 18:27.24 | lvlinux | oh |
| 18:27.26 | drmessano | So 50% more that Flowroute.. out.. If you split the calls 50/50 it's cheaper to use Flowroute |
| 18:27.39 | lvlinux | well i wonder where i got the 0.0045 from... |
| 18:28.14 | drmessano | They do have volume pricing, which is cheaper than Flowroutes PUBLISHED rates.. but Flowroute offers volume discounts as well |
| 18:29.07 | lvlinux | https://www.twilio.com/sip-trunking/pricing |
| 18:29.11 | drmessano | Whats interesting is that even 15m+ calls per month their outbound rate is STILL higher than Flowroutes publushed rates |
| 18:29.35 | lvlinux | that link says origination is 0.0045/min. ??? |
| 18:29.47 | drmessano | https://www.twilio.com/voice/pricing |
| 18:29.56 | lvlinux | yes but that's using the API |
| 18:30.50 | drmessano | LOL |
| 18:31.14 | lvlinux | ??? |
| 18:31.17 | lvlinux | am i missing something? |
| 18:31.20 | *** join/#asterisk dakudos (~kados@c-73-243-248-133.hsd1.co.comcast.net) |
| 18:31.58 | drmessano | "Starting at.." |
| 18:32.04 | drmessano | Their rates are variable |
| 18:32.15 | drmessano | If you look at the spreadsheet, it makes much more sense |
| 18:34.45 | drmessano | I would wager to say the pricing for their API calls is closer to the average |
| 18:35.15 | lvlinux | i figured the "starting at" meant for US local |
| 18:35.21 | lvlinux | i'll take a look at the CSV |
| 18:35.50 | drmessano | Nope |
| 18:36.30 | drmessano | Like literally it's .014 to call my work number |
| 18:36.35 | drmessano | and .007 to call my parents number |
| 18:36.43 | drmessano | They're both off the same exchange |
| 18:37.16 | lvlinux | yikes |
| 18:37.32 | lvlinux | yeah i don't like that---i like plain and clear rate models |
| 18:37.47 | lvlinux | but as far as the service, how has it been? any issues? |
| 18:39.01 | drmessano | Never heard a problem with it |
| 18:39.17 | drmessano | I know people that use the API for stuff |
| 18:39.39 | drmessano | It just doesnt make sense to use it as a traditional ITSP when there's many others |
| 18:39.42 | drmessano | Like Flowroute |
| 18:40.15 | lvlinux | well i was thinking about using Twilio for a backup and messing with the API maybe in the future. (Don't need it right now) |
| 18:40.31 | drmessano | That's not a bad option |
| 18:40.45 | lvlinux | I use Anveo primarily, but also DIDLogic, Flowroute, Voip.ms, and Vitelity |
| 18:41.19 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 18:41.26 | drmessano | I use Flowroute.. have for years.. I set up VoIP.ms a few years back as a backup |
| 18:42.06 | drmessano | Twilio seems to be popular and have lots of infrastructure. That makes it compelling backup to me |
| 18:42.53 | lvlinux | i may go ahead and sign up then. The SRTP option looks good too. |
| 18:42.55 | drmessano | Flowroute is also kinda easy to get others to use |
| 18:43.18 | drmessano | I dont sell phone service.. but I do sell and support the hardware |
| 18:43.28 | drmessano | Its nice to be able to put them onto Flowroute |
| 18:43.31 | drmessano | Self-service and all |
| 18:43.32 | lvlinux | Me too. and Flowroute doesn't proxy media, which is nice. |
| 18:43.47 | lvlinux | (IMHO) |
| 18:44.19 | drmessano | Look, when Flowroute is down.. BIG stuff is typically down |
| 18:44.30 | lvlinux | yeah |
| 18:44.40 | drmessano | Like the stuff that I would need in place to actually get the calls to Flowroute anyway (of they had proxies) |
| 18:45.06 | drmessano | Like Level3 sort of outages |
| 18:45.24 | drmessano | So.. not proxying the media is awesome |
| 18:45.31 | lvlinux | yup |
| 18:45.51 | drmessano | I dont want to be down because some idiotface typed rm instead of rem into a console |
| 18:48.39 | lvlinux | lol |
| 18:48.56 | lvlinux | Well looks like I'm gonna have to give up on this stupid gtalk problem... |
| 18:49.48 | drmessano | What is the need? |
| 18:49.56 | drmessano | Google is pretty close to killing XMPP for good |
| 18:50.27 | drmessano | If you have prosody, seems like you have a solution |
| 18:53.14 | lvlinux | i do have a working solution with prosody, but I'd still like to be able to connect directly to google to send messages to gtalk |
| 18:54.15 | drmessano | That's really almost extinct |
| 18:54.31 | lvlinux | not really---the messages go into Hangouts |
| 18:55.00 | drmessano | Not exactly |
| 18:55.33 | lvlinux | how so? it works for me with other things like bitlbee |
| 18:57.19 | drmessano | Many accounts don't have XMPP anymore. Many accounts have XMPP and can't send/receive voice calls with Asterisk because they're opted in for Hangouts. Many messages from XMPP <> Hangouts are simply failing. The level of reliability is 0 at this point and Google is not actively supporting XMPP.. it's just "Not turned off yet" |
| 18:58.28 | drmessano | We know every single time XMPP fails because we get flooded with complaints on IRC about "Google Voice stopped working.. I cant auth.. WTF" |
| 18:58.59 | drmessano | It's on shady life support that's connected to sparking UPS sitting in a puddle of Google Urine |
| 18:59.41 | lvlinux | lol |
| 19:01.03 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 19:01.36 | lvlinux | hmmm, so it just depends on when someone created their google account whether or not XMPP is enabled I guess. I use hangouts and GTalk both but I guess the accounts i communicate with via XMPP-to-Hangouts all just ahppen to have XMPP enabled... |
| 19:02.24 | drmessano | It's much more than that |
| 19:02.46 | drmessano | anymore = stopped working |
| 19:03.00 | drmessano | New accounts are hit and miss |
| 19:03.10 | drmessano | Some are just unreliable |
| 19:04.33 | drmessano | I think it's literally when a server dies that's handling the XMPP for a group of accounts, it's not replaced.. and if some inter XMPP<>Hangouts connector dies somewhere, it may not be replaced.. and when XMPP has died for some users, it's been down for days before google resolved it |
| 19:04.39 | drmessano | Life support |
| 19:05.52 | lvlinux | well, i guess i'll just ditch it for this server... |
| 19:06.12 | lvlinux | takes a hatchet and heads off towards xmpp.conf... |
| 19:06.52 | Samot | cranks Volbeat |
| 19:07.39 | drmessano | Whats sad |
| 19:07.49 | drmessano | Is that we had a somewhat working implementation in Asterisk |
| 19:08.12 | drmessano | Then someone went and wrote an awesome implementation |
| 19:08.20 | drmessano | and Google decided to start chopping shit lol |
| 19:09.32 | lvlinux | yeah |
| 19:09.44 | lvlinux | i guess the reason it worked a few minutes ago matched up with all you said... |
| 19:10.13 | drmessano | Theres one guy that comes in here, pretty regular |
| 19:10.22 | drmessano | He's the canary |
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| 19:10.27 | drmessano | Others follow |
| 19:10.44 | drmessano | But he's the first one "I can't get Google Voice to work" |
| 19:11.08 | lvlinux | i always use Yate for GV |
| 19:11.15 | file | remembers when someone tried to run a call center off Google Voice accounts |
| 19:11.21 | drmessano | HAH |
| 19:12.04 | drmessano | That's a lot of CAPTCHAs |
| 19:13.23 | lvlinux | lol |
| 19:13.30 | lvlinux | that's funny |
| 19:14.42 | drmessano | I believe it |
| 19:15.04 | lvlinux | "I'm sorry boss but it's not my fault that phones are down---it's Google, our SIP provider..." lol |
| 19:15.06 | drmessano | "How many X100P cards can I put in a chassis?" |
| 19:15.19 | drmessano | ^ Thats happened |
| 19:15.22 | lvlinux | lol |
| 19:15.48 | [TK]D-Fender | I ... remember those days... |
| 19:15.51 | lvlinux | how many X100p cards fit in a garbage can lol |
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| 19:21.44 | drmessano | lvlinux: All of them |
| 19:22.01 | drmessano | Its an interesting phenomon |
| 19:22.21 | drmessano | If you keep dropping x100p cards into the trash.. the trash gets bigger on the inside |
| 19:22.23 | igcewieling | lvlinux: the same as the number of jabberwockys which can fit in a garbage can. |
| 19:22.24 | drmessano | Like a TARDIS |
| 19:22.54 | lvlinux | lol |
| 19:22.56 | igcewieling | I think asking how many x100p clones can fit in a garbage can. REAL x100p has its problems, but was not a bad card. |
| 19:23.52 | lvlinux | yes i should have been clearer. put the real ones on the display shelf and the clones in the garbage. |
| 19:27.31 | drmessano | I cant imagine either stacks up against even a newer model single-port clone card at this point |
| 19:27.52 | drmessano | I played with an OpenVOX for some time and I was quite happy with it |
| 19:28.24 | drmessano | FXS isn't difficult to do.. FXO seems to be problem child |
| 19:32.17 | igcewieling | drmessano: the disadvantage with other cards is you have to pay almost as much for 4 ports as you do for 1 port. |
| 19:33.30 | drmessano | Depends.. I can get an OpenVOX single port for $100. If you're talking about Digium or Sangoma cards, the buy-in for one port is obviously much different |
| 19:34.21 | drmessano | That largely depends on need |
| 19:34.45 | lvlinux | I only buy Sangoma anymore. Every one I've used has been rock solid. |
| 19:35.08 | drmessano | I wouldn't buy a chinese PRI card.. But I would buy a chinese analog card |
| 19:35.19 | drmessano | 4 ports or less* |
| 19:35.32 | drmessano | If I needed more analog ports, I would buy a name brand |
| 19:36.08 | drmessano | The data doesn't really suggest the chinese are bad at the smaller port configurations on the analog side |
| 19:36.41 | drmessano | Above that it starts to get weird |
| 19:36.45 | [TK]D-Fender | I would sooner just buy a SIP gateway device |
| 19:36.56 | drmessano | ^ That too |
| 19:37.19 | [TK]D-Fender | Get rid of the exposed circuit risk, shoddy knock-off factor, hardware dependence, etc |
| 19:37.33 | [TK]D-Fender | Fuck it the used stuff is super cheap to buy |
| 19:37.44 | lvlinux | i use both, just depends on the particular installation. (but i always buy used) |
| 19:37.45 | [TK]D-Fender | especially from those who already bought it and moved to pure voip |
| 19:38.24 | drmessano | Getting people off of analog is always so nice |
| 19:38.34 | drmessano | Especially when you drop in an ATA and all their phones work |
| 19:38.42 | drmessano | so it's transparent |
| 19:38.47 | lvlinux | i have a couple of old MultiTech MultiVoIP 4 port FXO/FXS boxes. They are a royal pain to setup with the ancient windows only software but they work very very well. |
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| 20:01.54 | Micc | anyone know where I can find the ringing settings for indications.conf that would sound like the ringing in the matrix? |
| 20:16.54 | lvlinux | does cdr_adaptive_odbc.so depend on cdr_odbc.so? |
| 20:37.24 | igcewieling | lvlinux: It does not. I write cdrs using adaptive odbc and do not have a cdr_odbc.conf |
| 20:45.50 | lvlinux | k good. thanks |
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| 21:33.38 | lvlinux | anybody played with opus on v14? it's crashing my * for some reason (maybe I don't have it configured properly???) |
| 21:34.21 | igcewieling | nope. no reason to. |
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| 21:42.49 | lvlinux | nevermind, got it working |
| 21:49.26 | drmessano | nevermind then |
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