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| 00:55.05 | ketas | new asterisk is damn beast on resource usage |
| 00:57.20 | MaliutaLap | ketas: let's be fair here, it's no Zimbra |
| 00:57.41 | MaliutaLap | that is one huge resource hog |
| 01:04.53 | ketas | not loading million modules helped actually |
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| 02:13.45 | beebuu | hello,all |
| 02:14.56 | beebuu | is there anyone know how to setup a oversip server for asterisk? |
| 02:16.01 | beebuu | i can't find out with google |
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| 05:18.25 | ndnihil | howdy, anyone have thoughts on "Ignoring video stream offer because port number is zero"? |
| 05:19.00 | ndnihil | getting it no matter what client I use (as long as video is enabled), tried limiting the codecs, enabling all codecs, no change |
| 05:19.43 | ndnihil | system is raspbx/asterisk 11.18 |
| 05:21.03 | ndnihil | clients tried are sipdroid, linphone, ekiga, qutecom, simpleopal, and probably a couple others I didn't keep installed |
| 05:22.00 | ndnihil | all local LAN, no NAT |
| 05:22.07 | ndnihil | no firewalling between any devices |
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| 05:24.45 | ndnihil | google hasn't been much help, just 500 sites reposting the same mailing list thread with no resolution |
| 05:25.48 | ndnihil | one forum thread saying 11.2 was affected but 11.4 and later were not, but 11.18 here |
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| 05:47.53 | ndnihil | add csipsimple to the list |
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| 07:46.05 | denbeiren | Hi all,.. i am completely new to asterisk and i have a few practical questions before i start the learning curve :-) |
| 07:46.44 | denbeiren | I would like to install asterisk on my esx server,.. is this possible and what kind of hw would it need to run ? |
| 07:47.20 | denbeiren | i would have 1 or 2 landlines, combined with 1 voip account (for my securitycam/doorbell) |
| 07:47.39 | denbeiren | possibly integrate skype |
| 07:48.27 | denbeiren | if possible enable callforwarding to cellphone if home phone isn't answered |
| 07:48.41 | denbeiren | doorbell call to cellphone is noone is home |
| 07:49.15 | denbeiren | possibly some more small things,.. but this would be the basics |
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| 11:44.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 12:03.01 | Voyage | Hi |
| 12:03.36 | Voyage | Can asterisk or any pbx technology (I am non technical and unaware to use the right terminologies) make automated calls to a list of numbers with recorded messages? |
| 12:04.21 | wdoekes | asterisk can, and there is more than one way to do it |
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| 12:05.00 | wdoekes | one possibility is to use spool files for the call setup (origination) |
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| 12:05.27 | wdoekes | for playing recorde messages, sou set the other end of the call to a Local channel that answers and plays the recording |
| 12:05.57 | wdoekes | s/recorde/recorded s/sou/you |
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| 12:29.23 | Voyage | wdoekes, hm |
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| 12:37.43 | Voyage | wdoekes, Want to have an easy way to a) make one by one automated calls to a list of numbers with a recorded message b) if the user presses a number (say '1') I can be able to speak with him c) regardless of "b", I can always step in the call and speak in the middle of call. d) there is no hard phone in the scenario (only soft phone) |
| 12:45.26 | Voyage | wdoekes, what about sms? Can I do automated sms too? |
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| 14:21.53 | wdoekes | Voyage: see ~book for more information about building it; as for sms, it's easier to send sms using a dedicated sms provider through a HTTP API |
| 14:21.56 | wdoekes | good luck |
| 14:28.34 | Voyage | wdoekes, I want my own pbx/server. That would be cheaper. no? |
| 14:29.12 | WIMPy | Cheaper than what? |
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| 14:35.22 | zaf | anyone know of any good android/ios apps for remote workers to check/send voicemail, take incoming calls, etc? other than a softphone? |
| 14:36.35 | [TK]D-Fender | zapata: If it isn't a soft-phone .. then it's an ACTUAL PHONE call. |
| 14:36.41 | [TK]D-Fender | zaf: ^ |
| 14:37.04 | zaf | [TK]D-Fender, they want to do things like turn on/off forwarding of their calls, , etc |
| 14:37.31 | [TK]D-Fender | That part could be through some other interfeace, but call = softphone |
| 14:37.45 | Voyage | WIMPy, cheaper than usual sms services that charge per sms |
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| 14:38.01 | zaf | yeah, i guess i should have said "other than a generic softphone". something that talks to ARI/AMI I guess |
| 14:38.22 | Voyage | zaf, we can make an app for you if you dont find one |
| 14:38.58 | WIMPy | Voyage: you still need to send thme somewhere. And a dedicated app will be much more usefull. |
| 14:40.03 | Voyage | WIMPy, hm do you know any providers that you can suggest? |
| 14:40.27 | WIMPy | Google for "bulk sms". |
| 14:40.50 | Voyage | WIMPy, ya but that shows many generic results. What specific keyword do I need? |
| 14:41.36 | WIMPy | The one that doesn't make the results generic. Whatever that means. |
| 14:42.07 | Voyage | bulk sms also shows softwares that do that. |
| 14:42.11 | Voyage | but ok, i will try |
| 14:42.32 | WIMPy | You probably want that as well anyway. |
| 14:42.37 | Voyage | can anyone recommend a outgoing call and sms provider? voip? |
| 14:42.42 | Voyage | WIMPy, ok |
| 14:43.56 | [TK]D-Fender | zaf: Nope. |
| 14:44.33 | [TK]D-Fender | zaf: Audio doesn't "just happen". You need a channel driver... and that leads to "soft-phone" |
| 14:45.12 | zaf | [TK]D-Fender, a lot of 3rd party systems just bridge calls to your cell# for you |
| 14:45.32 | zaf | that's what this customer is asking for |
| 14:45.39 | [TK]D-Fender | that is your cell getting a call via the cellular network |
| 14:45.41 | WIMPy | What's stopping you from doing the same? |
| 14:45.55 | zaf | WIMPy, something to control that from the cell |
| 14:45.56 | Voyage | WIMPy, oh no. 4.88 cents per sms. Too costly. |
| 14:45.56 | [TK]D-Fender | Which has nothing to do with any software on your cell itself |
| 14:45.58 | [TK]D-Fender | it's a dumb call |
| 14:46.19 | [TK]D-Fender | Voyage: So do you see an ITSP that has it at a price you like? |
| 14:46.42 | Voyage | I just checked bulksms.com |
| 14:46.46 | Voyage | looking for more |
| 14:46.48 | zaf | they want to fire up an app and hit a button that says "i ain't at the office, send my calls here" |
| 14:47.17 | WIMPy | Voyage: Look for prepaid options. |
| 14:47.25 | [TK]D-Fender | Then you just need some PSTN connectivity on your server... which you probably have just to get an incoming call anyway |
| 14:48.12 | WIMPy | zaf: Just give them an URL to open in their browser. |
| 14:48.49 | Voyage | WIMPy, [TK]D-Fender dont the usual voip providers give sms facility too? I heard voip calls are very very very cheap |
| 14:49.09 | [TK]D-Fender | Voyage: They can be. As for SMS it depends |
| 14:49.16 | file | more VoIP providers are doing SMS but it depends on where the DIDs are, and some cost money |
| 14:49.18 | [TK]D-Fender | Voyage: Do not assume one service is related to the other. |
| 14:49.27 | WIMPy | SMS is not VOIP. And VOIP providers are usually not cheap. |
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| 14:52.08 | igcewieling | Ah, the traditional SMS discussion. That brings back memories. |
| 14:53.05 | ketas | i wonder if you can do dialup over ulaw |
| 14:53.27 | WIMPy | ketas: That's what you usually do. |
| 14:53.57 | ketas | would be fun to try... |
| 14:54.53 | WIMPy | If you're talking about using RTP, then there's no chance with a modem. With rata dadaption it actually works, but is horribly slow. |
| 14:55.02 | Voyage | ketas, sorry? dialup over ulaw? |
| 14:55.14 | ketas | well, fax works? |
| 14:55.34 | WIMPy | Sometimes. If you're lucky. |
| 14:55.43 | Voyage | [TK]D-Fender, WIMPy hm |
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| 14:55.56 | ketas | Voyage: yes! |
| 14:56.41 | WIMPy | LOL |
| 14:57.29 | WIMPy | I just see that the classical auto-attendant appliance vendors do VOIP versions now. Who do they think they can get with that idea? |
| 14:57.33 | Voyage | ketas, can you explain? |
| 14:59.32 | ketas | Voyage: that can't be too hard to understand? |
| 14:59.45 | ketas | i was wondering if it would actually work |
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| 15:05.27 | igcewieling | wondering what might actually work? |
| 15:05.57 | *** join/#asterisk mXr (mxr@84-112-233-130.dynamic.surfer.at) |
| 15:06.08 | mXr | good day :) |
| 15:06.44 | mXr | anyone here who might be able to help me with a weird problem regarding asterisk (13.5.0) sending RE-invites for an established t.38 call with audio sdp? |
| 15:07.26 | mXr | pretty much this https://issues.asterisk.org/jira/browse/ASTERISK-21045 |
| 15:07.42 | mXr | except, its supposed to be fixed already, .. |
| 15:08.06 | mXr | and checking thru the source, this patch https://issues.asterisk.org/jira/secure/attachment/46720/dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch is definitely applied |
| 15:08.09 | igcewieling | mXr: I believe reinvites are not supported in Asterisk w/T.38. |
| 15:08.23 | mXr | ok but why is it sending one then :P |
| 15:08.23 | igcewieling | file file: file! |
| 15:08.35 | igcewieling | file might know for sure. |
| 15:08.36 | mXr | im looking at a pcap from about half an hour ago.. a long fax |
| 15:08.41 | mXr | file? |
| 15:08.44 | file | in meeting, will answer soon |
| 15:08.47 | mXr | oh right |
| 15:08.52 | igcewieling | mXr: turn off reinvies |
| 15:09.06 | mXr | as in, canreinvite=no? |
| 15:09.25 | igcewieling | mXr: depends on your asterisk version. It should be obvious in sip.conf.sample |
| 15:09.26 | mXr | (further info, the other side is a linksys spa 112) |
| 15:09.33 | mXr | 13.5.0 |
| 15:09.40 | mXr | checking |
| 15:09.44 | ketas | igcewieling: a dialup |
| 15:09.51 | igcewieling | ketas: never going to happen. |
| 15:10.11 | igcewieling | an ULAW call uses around 80Kbps. Modems top out at 56K. |
| 15:10.22 | file | okay, lemme see here |
| 15:10.23 | igcewieling | sorry, a SIP ULAW call, not ULAW. |
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| 15:10.42 | mXr | hmm |
| 15:10.44 | WIMPy | Off course it works. It worked all the time. IF uyou use TDM. |
| 15:10.45 | file | reinvites are received or sent for T.38 to do the switch-over, can't stop it |
| 15:10.52 | mXr | yes no |
| 15:10.53 | ketas | igcewieling: i mean dialup over ulaw of course |
| 15:10.54 | mXr | thats not the issue |
| 15:11.01 | file | further information such as the full SIP log and console output is needed |
| 15:11.04 | mXr | file: im switching to t.38 properly, for pretty much exactly 300 seconds |
| 15:11.05 | igcewieling | file: should reinviews work with T.38? |
| 15:11.09 | mXr | then the asterisk side sends a reinvite with alaw |
| 15:11.11 | WIMPy | ketas: Your question is far too vague. |
| 15:11.18 | mXr | for no reason i can think of |
| 15:11.19 | ketas | how so? |
| 15:11.24 | igcewieling | ketas: oh. wow, that's funny. |
| 15:11.27 | file | when the fax session ends Asterisk can attempt to reinvite back to audio |
| 15:11.44 | mXr | well, it definitely doesnt end *properly* |
| 15:12.04 | mXr | btw, the use case (i didnt say that sorry) is to send a fax from a physical fax machine connected to the linksys spa 112 |
| 15:12.06 | igcewieling | file: my memory is worse than I thought. I was sure you told me to not try directmedia with T.38 because it should not be expected to work. |
| 15:12.07 | mXr | over the asterisk |
| 15:12.11 | WIMPy | ketas: You don't say anything about what kind of dialup or where the ulaw is going. |
| 15:12.23 | mXr | which means the spa is doing the initial t38 re-invite and all of that works |
| 15:12.29 | igcewieling | WIMPy: I think he means "modem over sip" |
| 15:12.37 | mXr | BUT ~300 seconds later, asterisk decides to bork it up by sending a re-invite himself |
| 15:12.38 | ketas | WIMPy: i just wondered if anything could work, just for fun |
| 15:12.49 | WIMPy | igcewieling: Maybe |
| 15:12.52 | file | without a SIP log and console output it's all guessing |
| 15:13.08 | WIMPy | ketas: And I gave you the answer: It depends. |
| 15:13.19 | igcewieling | Our solution to T.38 with Asterisk is to not use it with Asterisk 8-| |
| 15:13.20 | mXr | ah ad console output, im also getting something super weird |
| 15:13.28 | WIMPy | mXr: Could that be on session-timer expiry? |
| 15:13.34 | ketas | WIMPy: well i continued to talk with others |
| 15:13.36 | mXr | [2015-10-06 16:37:43.604] ERROR[17091][C-0000811e] res_rtp_asterisk.c: Received SSL traffic on RTP instance '0x7f07b4484718' without an SSL session |
| 15:13.42 | mXr | i have no f*n clue how that is related |
| 15:13.45 | mXr | i have tlsenabled=no |
| 15:13.52 | ketas | WIMPy: if i have more free time, i could see if i could do that :P |
| 15:13.54 | mXr | and i dont see any reason why the spa112 would send anything with tls either |
| 15:14.01 | mXr | WIMPy: well; i have set session-timer to refuse |
| 15:14.08 | WIMPy | ketas: But you didn't narrow your question. |
| 15:14.16 | mXr | because this problem existed in earlier asterisk versions, see the bug ticket i linked |
| 15:14.32 | ketas | WIMPy: to someone or what i actually wanted to do? |
| 15:14.52 | WIMPy | ketas: What you wnat to do. |
| 15:16.53 | mXr | btw, i have directmedia=no in my sip.conf |
| 15:17.04 | mXr | i also had canreinvite still but that should have had no effect anymore |
| 15:21.51 | Voyage | ketas, are you saying that I can send sms via voip ? |
| 15:22.03 | Voyage | ketas, I am no a techincal person |
| 15:22.24 | [TK]D-Fender | Voyage: I already told you that. Depends on your provider |
| 15:22.34 | ketas | i don't think i was the one who talked about sms |
| 15:23.27 | Voyage | cant find a cheap sms provider for united states |
| 15:23.40 | ketas | start using ip |
| 15:23.42 | ketas | :) |
| 15:23.47 | Voyage | ketas, ip? |
| 15:23.51 | Voyage | what do you mean? |
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| 15:24.06 | ketas | but i bet not everyone has smartphone yet |
| 15:24.17 | Voyage | ketas, what do you mean by ip? |
| 15:24.31 | Voyage | ketas, ip address? |
| 15:25.11 | ketas | like ip for transmitting messages |
| 15:25.31 | ketas | hell, i still use sms myself |
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| 15:30.31 | igcewieling | SMS is like Xerox, the term has been misused so much it doesn't really mean anything anymore. |
| 15:31.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:32.50 | [TK]D-Fender | Don't know about USA, but Xerox isn't used as a synonym for photocopy in Canada... |
| 15:33.05 | igcewieling | [TK]D-Fender: interesting. |
| 15:33.13 | [TK]D-Fender | However Kleenex reigns supreme in place of tissue paper. |
| 15:33.21 | igcewieling | ah, I was just going to ask that. |
| 15:33.39 | [TK]D-Fender | \o/ brand recognition beyond dictionary awareness |
| 15:34.18 | [TK]D-Fender | In Quebec there is one particularly retarded one the french have which is literally calling a camera a Kodak. |
| 15:34.35 | [TK]D-Fender | That one grates me.... |
| 15:35.07 | *** join/#asterisk saint_ (~saint_@c-71-59-68-185.hsd1.nj.comcast.net) |
| 15:37.03 | igcewieling | "sip trunk!" |
| 15:38.42 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:38.56 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:41.04 | Voyage | ketas, can you please elaborate? can I send sms via ip/voip? |
| 15:41.21 | Voyage | igcewieling, ^ |
| 15:41.38 | igcewieling | Voyage: Indeed. Just sign up with a provider offering that service. |
| 15:41.42 | WIMPy | Voyage: Yes, maybe. |
| 15:42.15 | Voyage | igcewieling, you mean I want to signup with a provider that provides xerox and I can send sms? |
| 15:42.20 | igcewieling | Vitelity supports "SMS" which you can use a standard Jabber client to interface with. I'm sure many, many others offer it too. |
| 15:42.25 | igcewieling | Voyage: Indeed! |
| 15:42.33 | ketas | xerox over pigeon |
| 15:42.35 | Voyage | igcewieling, and it will be cheap as vopi? |
| 15:43.00 | igcewieling | Voyage: cheap is a subjective term. I pay under $2/month for the telephone number + usage. |
| 15:43.19 | igcewieling | I almost never use the "SMS" service anymore. |
| 15:43.43 | Voyage | igcewieling, I am getting 3 cents per sms to United states. I need like 0.3 cents |
| 15:45.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:45.19 | ketas | hah, i talked about my plan to use dialup over voip to someone i know... he wanted to try that once |
| 15:45.56 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 15:45.57 | igcewieling | ketas: people try that all the time. |
| 15:46.01 | Kobaz | anyone need an extra astricon ticket |
| 15:46.02 | igcewieling | the fail, but they try. |
| 15:46.33 | ketas | that is, connect analog modem to ata, then use another modem with ata as second party... or use something that softwarely emulates "modem" |
| 15:48.07 | ketas | i understand that problem is that that audio channel is mainly meant for human speech? |
| 15:48.25 | WIMPy | Voyage: If you should ever find a price like that it will probably only deliver one out of ten messages. |
| 15:49.27 | WIMPy | ketas: That's not the problem. The PSTN uses µlaw/alaw as well. It's mainly about jitter or other timing issues. |
| 15:49.35 | ketas | hmm |
| 15:50.20 | WIMPy | And low quality analog parts, off course. |
| 15:50.21 | ketas | like, the issues that didn't exist in analog phone networks? |
| 15:50.36 | ketas | so modem doesn't work well |
| 15:50.40 | WIMPy | Exactely. |
| 15:51.15 | WIMPy | More likely not. |
| 15:51.17 | ketas | and all atas are also meant only for you talking there... not machines |
| 15:51.20 | *** join/#asterisk bkruse (~Adium@74.51.115.100) |
| 15:51.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:51.47 | Kobaz | ketas: iaxmodem |
| 15:52.14 | Kobaz | ketas: wrong |
| 15:52.34 | ketas | eh? |
| 15:52.34 | Kobaz | ketas: i use t38 on ATA's all the time to hook them up to faxes |
| 15:52.51 | ketas | well, fax is kind of different, isn't it? |
| 15:52.58 | Kobaz | <ketas> and all atas are also meant only for you talking there... not machines |
| 15:52.59 | WIMPy | That's not modem any more. |
| 15:53.02 | igcewieling | ketas: T.38 is totally different. |
| 15:53.04 | ketas | i know that people put faxes on atas |
| 15:53.15 | Kobaz | yoiu said "all ata" and "only for talking" |
| 15:53.18 | ketas | but for dialup, there's no need |
| 15:53.23 | WIMPy | Yes, it works sometimes. |
| 15:53.36 | ketas | Kobaz: well, mostly? |
| 15:53.45 | Kobaz | i'm just saying one case proves the all isn't an all |
| 15:53.53 | Kobaz | works 100% for me |
| 15:54.00 | Kobaz | not even mostly |
| 15:54.01 | igcewieling | trying modem over SIP is a form of mental masturbation. Might be kind of fun, but doesn't accomplish anything. |
| 15:54.33 | ketas | only gives good feeling? |
| 15:54.43 | Kobaz | i think i'm getting jaded like [TK]D-Fender |
| 15:57.09 | [TK]D-Fender | Some people can't handle my level of realism... |
| 15:57.31 | Kobaz | perhaps |
| 15:58.06 | Kobaz | all, every, and always are usually one of the easest things to disprove |
| 15:58.27 | Kobaz | and also never, and none |
| 16:01.26 | [TK]D-Fender | #betheexception |
| 16:02.19 | Kobaz | #logicftw |
| 16:05.25 | Voyage | WIMPy, hm anysolutions? |
| 16:05.47 | WIMPy | Voyage: Find more money |
| 16:06.34 | Voyage | win its a bulk sms. Needs to be cost effective |
| 16:07.11 | Voyage | WIMPy, ok, what might be a rough per month unlimited calls to united states decent voip package would be? |
| 16:07.47 | WIMPy | There's no such thing as unlimited. |
| 16:08.42 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 16:10.22 | [TK]D-Fender | WIMPy: Sure there is. I'm exposed to human stupidity all the time..... |
| 16:11.13 | WIMPy | Ok, but not in contracts. |
| 16:13.09 | *** join/#asterisk doome_ (~doome@77-234-67-65.pool.digikabel.hu) |
| 16:13.33 | igcewieling | Voyage: The USA is far behind Europe in many telecommunications things. |
| 16:14.04 | WIMPy | Not any more. |
| 16:14.09 | ketas | i wonder why people still use fax in parallel with internet |
| 16:14.15 | WIMPy | VOIP is used in Europe as well. |
| 16:14.17 | ketas | i know they do |
| 16:14.41 | Voyage | igcewieling, WIMPy ketas hm how much for a decent voip monthly unlimited calls? |
| 16:14.44 | glNito | Because they love having the hard "I just sent something" feeling |
| 16:14.47 | ketas | i only use asterisk because my isp cancelled analog phone service |
| 16:14.58 | ketas | :) |
| 16:16.23 | WIMPy | Voyage: Depends on how high you want the limit on unlimited to be. |
| 16:16.42 | Voyage | WIMPy, 8 hours constant calling by one phone line |
| 16:17.07 | igcewieling | Voyage: unlimited? I didn't know they existed. |
| 16:17.22 | Voyage | igcewieling, rough idea on 8 hours constant calling by one phone line? |
| 16:17.33 | igcewieling | For example Vitelity's "unlimited" plan is limited to 3,000 mins. |
| 16:17.42 | igcewieling | Voyage: a rough idea of what? |
| 16:18.55 | *** join/#asterisk nix8n82 (~AndChat56@2601:283:8201:d495:3126:38d6:a9ba:b8ab) |
| 16:19.40 | WIMPy | 5, 6 or 7 day/week? But as you can already guess from igcewielings answer, you will nedd a rather expensive kind of unlimited. |
| 16:20.43 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:22.35 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:32.31 | Voyage | igcewieling, on cost per month |
| 16:33.13 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 16:33.33 | igcewieling | Voyage: I am not a sales person. |
| 16:34.26 | igcewieling | 1) for my personal use I almost never use a phone. 2) I work for a carrier, but our prices and services don't match with what you need. |
| 16:35.25 | igcewieling | though, honestly, I doubt any carrier is a match with what you need. |
| 16:43.17 | *** join/#asterisk ttyp123 (~kunal@117.196.47.146) |
| 16:45.01 | Voyage | hm |
| 16:45.28 | Voyage | so no ideas on rough cost for 8 hours a day, 30days a month calling price |
| 16:45.36 | *** join/#asterisk velusunivers-sys (~Velus@cpc8-know14-2-0-cust829.17-2.cable.virginm.net) |
| 16:45.43 | Voyage | actually I am new here and dont even know what to buy |
| 16:45.49 | ndnihil | sounds like a spam operation? |
| 16:47.23 | velusunivers-sys | does anyone know much about the asterisk 2 billing api? |
| 16:48.01 | WIMPy | velusunivers-sys: The developers. |
| 16:48.11 | WIMPy | ~ask |
| 16:48.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:48.14 | velusunivers-sys | anyon in her |
| 16:48.35 | WIMPy | Even if you're not that likely to find support for a2b in here. |
| 16:52.00 | ndnihil | any thoughts on "Ignoring video stream offer because port number is zero"? |
| 16:52.02 | *** join/#asterisk mub (~jub@50-205-171-170-static.hfc.comcastbusiness.net) |
| 16:52.39 | ndnihil | present with multiple clients, any combination of codecs I've tried |
| 16:53.15 | ndnihil | and most of what google returns is a huge number of sites reposting the same unresolved mailing list thread |
| 16:54.13 | glNito | are you using NAT? |
| 16:54.20 | ndnihil | nope, all local |
| 16:54.34 | ndnihil | though I've tried multiple nat settings |
| 16:54.46 | ndnihil | since I saw mention of port randomization |
| 16:56.12 | ndnihil | there was a forum post that mention it was present in 11.2.x, but not prior or later |
| 16:56.18 | ndnihil | but it's 11.18 here |
| 16:57.11 | igcewieling | ndnihil: if you are not using video you can ignore that message |
| 16:57.27 | ndnihil | were I not trying to use video I'd do just that |
| 16:57.33 | ndnihil | but I'm trying to get video to work |
| 16:58.48 | igcewieling | ah. The fix your video phones 8-) what actual client are you trying to use? |
| 17:00.03 | ndnihil | tried sipdroid, csipsimple, linphone (both android and linux), qutecom, ekiga, simpleopal, and a couple others for testing |
| 17:02.58 | igcewieling | ah. The only ones I've seen are Polycom ones. |
| 17:03.44 | igcewieling | ndnihil: This is VERY important: Asterisk CANNOT transcode video. |
| 17:03.55 | glNito | : |
| 17:04.15 | glNito | ^ was supposed to go into vim |
| 17:04.21 | ndnihil | I'm not asking it to do any transcoding that I'm aware of |
| 17:04.29 | ndnihil | just arrange the connection |
| 17:07.01 | mXr | trying again, maybe someone with insight is online now... anyone got a clue why my asterisk 13.5.0 is sending a re-invite after almost exactly 300 seconds during a t.38 session, thus killing it? |
| 17:07.22 | mXr | very similar to this bug https://issues.asterisk.org/jira/browse/ASTERISK-21045 except THIS particular bug has been patched a long time ago |
| 17:11.24 | Katty | file: YOU |
| 17:11.49 | Katty | file: are to have an excellent bday!!!!!! |
| 17:11.57 | Katty | file: happy bday :> |
| 17:12.49 | igcewieling | mXr: sounds like it is time to create packet capture of the problem and hope someone here will help. |
| 17:15.24 | mXr | i have one; |
| 17:15.33 | mXr | i just dont really want to post it on a public website tbh |
| 17:15.41 | mXr | and i cannot find any software that would anonymize a sip pcap |
| 17:15.55 | mXr | except "sianto" which doesnt even compile :P |
| 17:15.57 | ndnihil | sed |
| 17:16.11 | mXr | in a pcap? |
| 17:16.23 | ndnihil | dump it to plaintext |
| 17:16.34 | *** join/#asterisk [373K] (~Adium@frontdoor.fiveninety.373k.net) |
| 17:16.38 | mXr | but from what i can tell, its really fairly obvious; something compells asterisk to send a re-invite after 300 secs even tho there is absolutely no reason to |
| 17:16.43 | ndnihil | it'll end up human readable at some point anyway |
| 17:16.45 | mXr | session-timers are set to refuse even |
| 17:17.28 | mXr | i could mail it if someone is willing to take a look at it; its not like super secret, i just dont want numbers/ips on the public web :) |
| 17:18.06 | WIMPy | They will be scanned anyway. All of them. |
| 17:18.15 | mXr | im fairly sure this is a genuine asterisk bug, tho |
| 17:18.27 | mXr | because of the weird ssl messages that follow the re-invite |
| 17:18.59 | WIMPy | Are you using TLS? |
| 17:19.02 | mXr | nope |
| 17:19.05 | mXr | tlsenable=no |
| 17:19.27 | mXr | i dont have immediate access to the other side (linksys spa) atm but im fairly sure nothing in regards to tls was ever touched/set there either |
| 17:19.45 | WIMPy | So where do the ssl messages come from? |
| 17:20.26 | mXr | [2015-10-06 16:20:33.910] ERROR[11657][C-00007e38] res_rtp_asterisk.c: Received SSL traffic on RTP instance '0x7f07b5495728' without an SSL session |
| 17:20.28 | mXr | asterisk debug console |
| 17:20.57 | WIMPy | SRTP? |
| 17:20.59 | mXr | well, wireshark does show that after the "unwelcome" re-invite, the spa is sending rtp packets with "unknown rtp version 0" |
| 17:21.08 | ndnihil | looks like one end is trying to SSL where the other isn't |
| 17:21.13 | mXr | so maybe it really does send srtp for some reason |
| 17:21.16 | mXr | but why would it |
| 17:21.30 | mXr | again, this is in the midst of a 0815 t.38 fax send, after ~exactly 300 seconds of it working perfectly |
| 17:21.40 | mXr | if the call is less than 300 seconds, it works fine |
| 17:22.07 | mXr | im not super-experienced in reading sdp but |
| 17:22.17 | mXr | i dont see anything in regards to srtp being requested by asterisk |
| 17:22.49 | vader- | Can anyone recommend a IP ATA that has an FXO port? I have a paging system which provides a dial tone over a POTS line and I need to interface our asterisk system which is all IP based so that when someone form their IP phone dials an extension it acts like it's dialing on the paging system. |
| 17:22.55 | ndnihil | rtpholdtimeout=300 |
| 17:23.04 | ndnihil | does increasing that impact behavior? |
| 17:23.16 | mXr | i would have to test that in ~1 hour |
| 17:23.35 | mXr | when i have someone in front of that stonage tech device again |
| 17:23.35 | mXr | aka fax |
| 17:23.45 | ndnihil | mine was set to 300 by default anyway |
| 17:24.11 | mXr | dont have the setting, |
| 17:24.19 | mXr | and sip show settings gives me |
| 17:24.20 | mXr | <PROTECTED> |
| 17:24.20 | mXr | <PROTECTED> |
| 17:24.21 | vader- | our asterisk box does have a TDM2400P board but it has all FXS ports, i guess there is no way to convert them so they don't send a dial tone |
| 17:24.34 | ndnihil | I just did a: grep 300 /etc/asterisk/* |
| 17:24.46 | mXr | yeah its set to 300 but commented out in the default sip.conf |
| 17:24.47 | WIMPy | vader-: No, you need the correct module. |
| 17:24.52 | ndnihil | ah |
| 17:25.03 | ndnihil | well, that could still be the default, even if commented out |
| 17:25.19 | ndnihil | maybe uncomment and increase |
| 17:25.30 | mXr | can try yeah |
| 17:25.43 | mXr | i do have rtpkeepalive=60 |
| 17:25.51 | mXr | for some reason |
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| 17:36.43 | mXr | http://i.imgur.com/0Sg0207.png this is the re-invite that asterisk sends after roughly 300 seconds |
| 17:38.24 | mXr | a 0815 voice call invite as far as i can tell |
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| 17:41.09 | WIMPy | never realized that numbers can have a language. |
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| 19:15.07 | *** join/#asterisk markl (~markl@d54C2C766.access.telenet.be) |
| 19:18.24 | markl | Hi All |
| 19:18.41 | markl | I'm having a strange asterisk problem, anyone available for help? |
| 19:18.56 | WIMPy | ~ask |
| 19:18.56 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:19.58 | markl | look like some process "hangs" sometimes a day |
| 19:20.26 | markl | when it occurs audio is interrupted and peers are going unreachable |
| 19:21.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:23.37 | WIMPy | So your Asterisk keeps running? |
| 19:23.39 | *** join/#asterisk airjump (~Thunderbi@p20030070CE3630148154F81020B6E6B1.dip0.t-ipconnect.de) |
| 19:23.45 | WIMPy | How do you resolve the issue? |
| 19:24.02 | markl | yes asterisk keeps running |
| 19:24.40 | markl | after a while, between 5-30 secs everything is normal |
| 19:25.11 | markl | looks like an network dropout |
| 19:25.26 | markl | except ping and other network services keep running |
| 19:25.35 | WIMPy | Yes, smells like some sort of I/O issues. |
| 19:25.56 | WIMPy | Are you recording calls? |
| 19:26.00 | markl | yes |
| 19:26.25 | WIMPy | Maybe your disk is too slow. |
| 19:28.06 | markl | It's a brand new virtual server |
| 19:28.11 | markl | how could I check? |
| 19:28.49 | WIMPy | You would see your RAM fill up. |
| 19:28.56 | WIMPy | Also check for I/O waiting time. |
| 19:29.59 | robmal | I think the virtual server is an issue, the clock there is far from perfect when it hits max allowed resources. |
| 19:31.01 | markl | we're checking the clock source with the cli command "timer test 100" |
| 19:31.35 | markl | doesn't indicates any problems |
| 19:33.06 | glNito | did you rule out packet loss? |
| 19:33.32 | markl | no packet loss measured on the interfaces |
| 19:33.42 | *** join/#asterisk airjump (~Thunderbi@p20030070CE3630148154F81020B6E6B1.dip0.t-ipconnect.de) |
| 19:34.31 | markl | if we record the rtp stream using tcpdump on the server the incoming audio from phones and sip trunk is perfect |
| 19:34.52 | markl | the audio which leaves the server is interupted suddenly in a call |
| 19:36.13 | markl | how should we monitor I/O waiting time? |
| 19:36.44 | WIMPy | top |
| 19:36.46 | WIMPy | vmstat |
| 19:38.38 | markl | looking at load, it almost never exceeds 1.00 (server has 2 cpu's) |
| 19:39.08 | *** join/#asterisk brian_373K (~brian_373@2607:fb90:212f:d4a:6bda:18cc:c3c5:b800) |
| 19:39.12 | brian_373K | We're looking for non-dialer users to beta test our platform. We'll be giving free termination for feedback. If you're interested, email info@373k.net or join #373K. |
| 19:39.20 | WIMPy | Where's the CPU time ging? |
| 19:39.56 | WIMPy | brian_373K: What kind od platform? |
| 19:41.13 | brian_373K | WIMPy: SIP termination |
| 19:41.49 | WIMPy | brian_373K: Still a rather vague description. |
| 19:42.04 | markl | cpu time at this moment: cs 495 us 2 sy 2 id 95 wa 0 st 0 |
| 19:42.29 | brian_373K | Ok, class 4? |
| 19:43.28 | brian_373K | We're just looking to run a little traffic and make sure it goes as smoothly as our testing has shown. |
| 19:44.08 | brian_373K | US Domestic Termination |
| 19:44.16 | brian_373K | WIMPy: that help? |
| 19:44.30 | WIMPy | Yes |
| 19:44.54 | brian_373K | Whew :) |
| 19:46.25 | robmal | Clients' Voice |
| 19:46.25 | robmal | avatar "Whoa!" |
| 19:46.25 | robmal | Patricia, New York City |
| 19:46.31 | robmal | Nice. |
| 19:46.40 | [373K] | when sales guys enter geekdomâ¦. film at 11 |
| 19:47.01 | brian_373K | Hey, at least i know know class 4 and 5 |
| 19:47.32 | WIMPy | doesn't |
| 19:48.32 | cj | o/ /o |
| 19:48.45 | brian_373K | WIMPy: we can probably do some consulting for you if you need it. Lol |
| 19:49.03 | brian_373K | We'll set up some free webinars with lots of PowerPoint |
| 19:49.27 | WIMPy | I'm pretty sure it's nothing I have to know in my domain :-) |
| 19:49.51 | brian_373K | Continued Ed Credits? |
| 19:50.05 | [373K] | WIMPy: Easiest explanation⦠Class 5: Touches the subscriber full of features like voicemail⦠Class 4: Raw traffic between carriers. |
| 19:52.54 | brian_373K | I'll go ahead and cancel your webinar then.... |
| 19:53.46 | WIMPy | Not that interesting to me anyway, I guess. I make less than one call per decade to the US. |
| 19:54.40 | SpeakerToMeat | Is there a way to set a chan sip "trunk" that uses encryption, so that I can provide/check the certificate and it wont work if cert changes? using tls and srtp (not zrtp) |
| 20:02.13 | robmal | BTW, 373K guys, it should be 374K to fit your slogan :-( |
| 20:03.28 | robmal | Also, only employee on LinkedIn is Kelvin = good job. |
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| 20:04.32 | [373K] | People actually use that site? |
| 20:04.54 | markl | WIMPy: I set up some vmstat logs, hope to see some changes if the problem occurs again |
| 20:05.11 | robmal | [373K]: That's where we get all the HR spam from, d'oh. |
| 20:06.48 | brian_373K | 373.2 didn't seem ideal |
| 20:07.32 | cj | Could someone take a look at https://wsg.colliertech.org/unclassified/for-%23rhel/yum-update.log with me? |
| 20:07.53 | cj | Starting at line 470, I get some debug info about failing dependency resolution |
| 20:08.30 | robmal | But one above is 374.15K :-/ |
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| 20:08.31 | cj | I'm thinking this comes from the installation of some third party packages that were required by imagemagick |
| 20:08.39 | cj | robmal: shall I xz? |
| 20:09.45 | robmal | Skip broken first. |
| 20:09.58 | cj | robmal: that doesn't work. tried. |
| 20:10.06 | cj | I did all of the things mentioned in the logs as well as... |
| 20:10.24 | robmal | So find a repo that serves the versions you need or download the rpms by hand. |
| 20:10.34 | cj | oh, shoot. this is not #rhel. sorry! |
| 20:10.40 | robmal | Ye ;-) |
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| 20:17.20 | WIMPy | BTW: What would Class 1 2 and 3 be then? Intergalactic, Interplanetary and International? |
| 20:18.15 | [TK]D-Fender | Insterstellar should come before intergalactic |
| 20:20.13 | WIMPy | Stars and planetary systems form galaxies. |
| 20:20.16 | glNito | Class 1 is Multi-Universal |
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| 20:20.34 | WIMPy | Like Sinix? |
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| 20:31.22 | brian_373K | 1-3 aren't commonly used terms anymore |
| 20:31.30 | brian_373K | But 1 was international |
| 20:31.55 | brian_373K | 2 was tandems interconnecting regions of AT&T |
| 20:32.30 | brian_373K | 3 was tandems connecting larger areas within a region of AT&T |
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| 20:59.41 | mXr | i think i solved my re-invite mystery from earlier, but i cannot confirm it yet |
| 20:59.56 | mXr | the call was routed to yet another asterisk - but THAT one is old, and didnt have that t38/reinvite patch yet |
| 21:00.08 | mXr | so MINE probably just relayed the info of falling back to audio.... instead of generating it itself |
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