| 00:01.51 | *** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta) |
| 00:03.41 | [TK]D-Fender | no |
| 00:03.59 | [TK]D-Fender | You also seem to like shoving quotes around everywhere |
| 00:04.08 | [TK]D-Fender | And no visible logic as to why |
| 00:04.27 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Expressions |
| 00:04.31 | [TK]D-Fender | Read your basics |
| 00:05.56 | Jack17 | exten => s,5,GotoIf($[${iWordsCount} > 1]?${EXTEN:1},1) |
| 00:06.19 | Jack17 | exten => n,GotoIf($[${iWordsCount} > 1]?${EXTEN:1},1) |
| 00:06.31 | Jack17 | tdo you think this will work |
| 00:12.15 | Jack17 | and how do i enable early offer in asterisk 12 |
| 00:12.28 | Jack17 | i want to be able to hear the dialer voice before the receiver answers |
| 00:12.35 | Jack17 | how can i make sure my aasterisk is sending that |
| 00:17.08 | [TK]D-Fender | Also makes no sense |
| 00:17.15 | [TK]D-Fender | you're trying to do AMD before BRIDGING |
| 00:17.35 | [TK]D-Fender | You can't be LISTENING and have early media |
| 00:18.57 | freexer | I have a google voice connection working. I am trying to look at something similar to sip show channelstats. Is there such a command for motif I guess it is. Currently if I do a sip show channelstats it shows just one entry. I'm assuming there are two channels though--one for my phone and one to googles servers? Am I thinking about that incorrectly? Thanks |
| 00:20.14 | [TK]D-Fender | Hit TAB at CLI |
| 00:20.21 | [TK]D-Fender | Do you see a motif menu? |
| 00:20.58 | freexer | no |
| 00:21.33 | [TK]D-Fender | There's you answer |
| 00:21.59 | freexer | ok so you can't see the channel to google? |
| 00:23.04 | [TK]D-Fender | please rephrase your question |
| 00:23.07 | freexer | i guess i can use wireshark |
| 00:23.11 | freexer | ty' |
| 00:23.36 | [TK]D-Fender | If CLI doesn't show you a command option then there's no monitoring it |
| 00:23.46 | freexer | thank you |
| 00:24.57 | freexer | TK there is an rtp option...just out of curiosity can I tell the available options for that by typing rtp help or something |
| 00:25.16 | freexer | i tried it but it doesn't work...got a feeling that is there but I don't know wtf i'm doing |
| 00:25.20 | [TK]D-Fender | rtp <TAB> |
| 00:25.28 | [TK]D-Fender | you don't jsut invent words |
| 00:25.36 | [TK]D-Fender | CLI has autocomplete for a reason |
| 00:25.57 | freexer | freaking sweet |
| 00:26.00 | freexer | thanks |
| 00:27.43 | freexer | oh shiznit...i'm a guru now |
| 00:27.44 | freexer | lol |
| 00:28.46 | Jack17 | [TK]D-Fender i can listen and heave early media |
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| 01:23.24 | Samot | Man this is a lot of work for end users not to hear an upstream provider messege. |
| 01:25.37 | *** join/#asterisk aandrew (foobar@gromit.mixdown.ca) |
| 01:25.42 | aandrew | hello everyone |
| 01:26.15 | aandrew | I'm wondering if anyone has software recommendations for asterisk-compatible fax software for iOS (e.g. something that will let me send a fax from an iPhone over a SIP conection) |
| 01:27.22 | Samot | Thats a very specific thing. Im not sure anyone does it. |
| 01:27.52 | Samot | If they do its probably for their own platform and guessing not free. |
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| 01:54.13 | [TK]D-Fender | That is crazy specific |
| 01:54.32 | [TK]D-Fender | and since you're dealing with a small computer .. who needs to limit themselves to SIP? |
| 01:54.55 | [TK]D-Fender | Send the file to your server and let the server do the work, not the phone |
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| 02:13.25 | Bhakimi | anyone knows how hard i can push the asterisk http ami module? |
| 02:13.52 | Bhakimi | i currently use telnet to connec tto the manager and i push about 200-300 requests a second, i like to switch those to the web service |
| 02:14.41 | [TK]D-Fender | I'm not sure why it'd be any different |
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| 02:32.39 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 03:12.19 | WIMPy | Added complexity can't make yhings better. |
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| 03:51.30 | aandrew | Samot: well I'm thinking groundwire is an amazing ios voip softphone. I'm looking for a nice voip softphax. :-) |
| 03:52.09 | Samot | How often are people sending faxes from their phones? |
| 03:52.25 | aandrew | admittedly not that often |
| 03:52.45 | Samot | So with so little demand and so many other options to do such a thing... |
| 03:53.16 | Samot | The only softfax client you will find will 99% be someone's custom design and platform. |
| 03:53.26 | aandrew | there arent' "so many other" options really. t.38 on a phone wouldn't be all that CPU intensive either |
| 03:53.44 | aandrew | my printer can fax through a little ata, I'm just trying to eliminate the paper |
| 03:53.50 | drmessano | Why dont you set up email to fax |
| 03:53.56 | Samot | ^ That |
| 03:53.56 | drmessano | Far easier |
| 03:54.02 | drmessano | SMTP is a better transport for it |
| 03:54.09 | Samot | Or the fact most digital fax platforms have GUIS |
| 03:54.15 | drmessano | Then let the SMTP to FAX on the PBX do the leg work |
| 03:54.32 | Samot | Fire up browser, login to portal, send fax as normal. |
| 03:54.57 | aandrew | that sounds like ass tbh |
| 03:55.23 | drmessano | ROFL |
| 03:55.23 | Samot | aandrew: Perhaps you're not taking into consideration something has to convert that document into something that is sent over SIP. |
| 03:55.33 | drmessano | So you need to scan it |
| 03:55.33 | aandrew | just trying ot leverage things I have now rather... having to add a web app and email-to-fax gateway and the resultant UI headaches ... |
| 03:55.43 | drmessano | What? |
| 03:55.47 | drmessano | .... |
| 03:55.51 | aandrew | hm? |
| 03:55.56 | drmessano | They send the email from their MAIL CLIENT |
| 03:56.05 | aandrew | I can pick a picture or pdf, convert to TIFF is not a big deal |
| 03:56.08 | drmessano | You can SCAN and EMAIL so easily on IOS |
| 03:56.18 | Samot | Then how do you send that TIFF over SIP? |
| 03:56.19 | drmessano | So youre going to write the app? |
| 03:56.23 | aandrew | drmessano: it's the "Fire up browser, login to portal, send fax as normal." that I'm referring to |
| 03:56.39 | drmessano | I never suggested that |
| 03:56.41 | aandrew | Samot: T.38 and SIP |
| 03:56.43 | aandrew | drmessano: no, you didn't |
| 03:57.12 | drmessano | Scanner pro from Readdle. Use it all the time. |
| 03:57.16 | Samot | aandrew: How do you send that TIFF over SIP? How do you get it into the data stream? |
| 03:57.24 | drmessano | Just add an email to fax gateway |
| 03:57.26 | drmessano | Pretty simple |
| 03:57.30 | aandrew | drmessano: yes, I do too. all my receipts get scanned with that |
| 03:57.41 | aandrew | Samot: you place a SIP call |
| 03:58.05 | aandrew | Samot: I'm not a stranger to asterick nor t.38, I'm just trying to determine if what I'm looking for already exists |
| 03:58.09 | Samot | How does the SIP call KNOW to use that document to send in the stream?! |
| 03:58.15 | drmessano | It doesnt exist |
| 03:58.16 | Samot | It does not. |
| 03:58.17 | aandrew | Samot: are you familiar with SIP and T.38? |
| 03:58.27 | Samot | Quite. |
| 03:58.34 | aandrew | ok, so I'm not sure I understand your question then |
| 03:58.41 | Samot | I do understand the question. |
| 03:58.56 | Samot | I'm pointing out all the things that have to be done for your app to exist. |
| 03:59.00 | Samot | And why it doesn't. |
| 03:59.09 | Samot | Because it's not worth it. |
| 03:59.15 | aandrew | sending the fax is another layer on top of a normal VOIP call that any softphone app can already do |
| 03:59.18 | aandrew | you're probably right |
| 04:00.11 | Samot | When it comes to determining if projects like this are worth the time, costs and resources; I generally am. |
| 04:00.24 | aandrew | heh |
| 04:00.45 | aandrew | I know it's been a while since I've hung around the asterisk channels, but I don't recognize your nick |
| 04:01.10 | aandrew | drmessano I definitely remember and there are a bunch of hte usuals |
| 04:01.41 | Samot | Well it's been at least a year or so. |
| 04:01.48 | drmessano | Longer |
| 04:01.51 | drmessano | Closer to 2 |
| 04:01.56 | Samot | Almost. |
| 04:01.59 | aandrew | *nods* I'm from the pre-asterisk-10 era :-) |
| 04:02.20 | Samot | I've been using Asterisk since 1.2 |
| 04:02.23 | aandrew | actually right around the zaptel-dahdi rename |
| 04:02.36 | drmessano | That was the 1.4 era |
| 04:02.58 | aandrew | yeah so I'm about as old as you (in asterisk time) |
| 04:03.22 | aandrew | what the hell else was going on then, it was before digium got big and before bkw left to do freeswitch and all that |
| 04:03.25 | aandrew | holy cow it's been a long time |
| 04:07.35 | *** join/#asterisk Cory (~Cory@unaffiliated/cory) |
| 04:19.14 | lorsungcu | im a hundred years old and invented the telephone |
| 04:19.18 | lorsungcu | get bent |
| 04:39.52 | aandrew | hm? |
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| 07:56.48 | *** join/#asterisk infobot (ibot@rikers.org) |
| 07:56.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:43.16 | ShuttleDuck | Would you consider a SIP 200 OK response with an expires parameter in the "To" header as malformed? I thought expires were only for contact header? |
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| 10:00.13 | Jack17 | Samot |
| 10:00.40 | Jack17 | exten => n,GotoIf($[${iWordsCount} => 1]?${EXTEN:1}) is this correct |
| 10:01.46 | Jack17 | iWordsCount - is internal variable for AMD |
| 10:01.53 | Jack17 | ? |
| 10:02.03 | Jack17 | it will not work in asterisk right |
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| 10:12.45 | Martin` | Jack17: does >= instead of => work? |
| 10:13.28 | Martin` | we lost him |
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| 10:42.49 | stefan27 | whenever you cant remember which works, you can always do negation outside like $[ ! 3 < 4 ] |
| 10:43.16 | stefan27 | if it's integers |
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| 11:49.22 | Jack17 | <PROTECTED> |
| 11:49.22 | Jack17 | <PROTECTED> |
| 11:49.33 | Jack17 | s |
| 11:49.54 | Jack17 | so iwords count not nternal varibla |
| 11:50.06 | Jack17 | internal variable cant be used right guys |
| 11:52.19 | Jack17 | Samot |
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| 12:36.15 | Samot | fWhat? |
| 12:40.17 | Jack17 | i want to use condition on amd |
| 12:40.29 | Jack17 | iWordscount cant be used right |
| 12:41.07 | Samot | What are you talking about? |
| 12:42.07 | Jack17 | while using amd detection . there is a variable called iwordscount |
| 12:42.18 | Jack17 | when it detect 1st word iworddcount is 1 secondd iwordcount=2 |
| 12:42.23 | Jack17 | can we use the irword count |
| 12:42.53 | Jack17 | exten => n,GotoIf($[${iWordsCount} => 1]?${EXTEN:1}) |
| 12:43.01 | Samot | How are you setting iWordsCount? |
| 12:46.12 | Jack17 | what do you mean |
| 12:46.27 | Jack17 | iwords count from amd |
| 12:46.27 | Samot | What makes you think that iWordsCount is an AMD variable? |
| 12:46.38 | Jack17 | i was asking if it is |
| 12:46.47 | Jack17 | ok can i use minword in amd.conf |
| 12:46.52 | Jack17 | instead of maxwords |
| 12:47.18 | Jack17 | minimum number of words |
| 12:47.23 | Samot | That's how many seconds you want a word to be. |
| 12:47.26 | Samot | Not how many words. |
| 12:47.39 | Jack17 | maximumNumberOfWords - Is the maximum number of words in a greeting |
| 12:47.45 | Samot | Yes. |
| 12:47.49 | Jack17 | the problem is |
| 12:47.52 | Samot | But there is no minimum version of it. |
| 12:47.56 | Jack17 | amd detect too longs words as notsure |
| 12:48.00 | Jack17 | and silent as notsure |
| 12:48.08 | Jack17 | i want to stop silent calls |
| 12:48.23 | Jack17 | i cant because silentcalls are THIS IS NOT SURE and tooo many words is THIS NOT SURE |
| 12:48.27 | Jack17 | same results in AMD |
| 12:48.46 | Samot | They you have your maximumWordLength should be adjusted. |
| 12:49.00 | Samot | What do you mean silent calls?! |
| 12:49.08 | Samot | You mean when the other side answers and is silent?! |
| 12:49.37 | Jack17 | <PROTECTED> |
| 12:49.37 | Jack17 | <PROTECTED> |
| 12:49.37 | Jack17 | <PROTECTED> |
| 12:50.09 | Samot | You haven't show what your actual AMD settings are. |
| 12:50.23 | Samot | So all of this is just pointless because that log could be output proper results. |
| 12:51.16 | Samot | Like I said to your "partner" this is a lot of work for a solution that shouldn't be a technical solution. |
| 12:53.47 | Samot | Of course. |
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| 12:54.42 | Samot | jack17: You realize your AMD solution limits your end users to HUMAN calls only? |
| 12:54.42 | jack17 | Samot silent is when like this |
| 12:55.04 | jack17 | <PROTECTED> |
| 12:55.23 | Samot | You haven't show what your actual AMD settings are. |
| 12:55.33 | jack17 | <PROTECTED> |
| 12:55.41 | jack17 | - AMD: Channel [SIP/1041-00000228]. Too long... |
| 12:55.45 | Samot | Stop pasting random log entries. |
| 12:55.48 | Samot | Show your AMD settings. |
| 12:56.03 | jack17 | ~pb |
| 12:56.03 | infobot | pastebin is, like, 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. |
| 12:59.45 | jack17 | https://gist.github.com/anonymous/403b14cad6fa49b64aff4c13b32500dd |
| 12:59.53 | jack17 | look same results on different scenarios |
| 13:00.13 | Samot | Is this a reason you are not showing what you are specifically are being asked for? |
| 13:00.40 | Samot | Is this a reason you are not showing what you are specifically being asked for? |
| 13:00.42 | jack17 | i will paste amd link |
| 13:00.53 | Samot | It's ONE LINE |
| 13:01.02 | Samot | AMD(SETTINGS HERE) |
| 13:01.07 | Samot | ^ WTF ARE THEY? |
| 13:01.25 | jack17 | https://gist.github.com/anonymous/39cbc062475932459bfb257cdd72ef1c |
| 13:01.46 | jack17 | man this has nothing to do with amd setting i believe |
| 13:02.04 | jack17 | because amd will return not sure when silence and when too many words detected will return notsure also |
| 13:02.13 | jack17 | :( |
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| 13:07.20 | Samot | Why are you only doing the check for 5 seconds? |
| 13:07.55 | jack17 | we can do it for 2 sec |
| 13:08.10 | jack17 | but i need to stop the first scenario i showed you |
| 13:09.31 | Samot | It's detecting too much silence? |
| 13:09.47 | jack17 | no silence will be NOTSURE as a result |
| 13:09.55 | jack17 | i want to stop silence |
| 13:10.15 | jack17 | dont allow silence calls to pass |
| 13:10.41 | Samot | Well a silent call means it can't tell if it's human or machine.. |
| 13:10.45 | Samot | Since it's silence.. |
| 13:10.57 | Samot | So yeah, NOTSURE is the proper response. |
| 13:11.16 | Samot | So then hangup NOTSURE results. |
| 13:11.26 | jack17 | ok but when its detecting a lot of words it reply not sure as well |
| 13:11.45 | jack17 | Samot you are right but same results occur when detecting many words |
| 13:12.17 | Samot | Like I said before, this is the wrong solution. |
| 13:13.53 | Samot | Outside of it being a technical solution for a non-technical problem (as I see it), it introduces a bunch of factors that impact your end users ability to successfully make calls. |
| 13:15.46 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:16.21 | Samot | Like I told your partner, follow-me, AMD is not 100%. It's going to return false positives on some of the calls. You can't avoid it. |
| 13:20.06 | Samot | I'm just trying to figure out what scenario happens that a user would call a destination that answers with pure silence. |
| 13:20.25 | Samot | Because that isn't a normal scenario. |
| 13:21.12 | [TK]D-Fender | s/partner/self/ |
| 13:21.22 | Samot | Yeah, I know. |
| 13:21.32 | Samot | I was being considerate. |
| 13:21.48 | [TK]D-Fender | I considered for a moment, then I reconsidered and NAH.... |
| 13:22.08 | Samot | This is just a horrible solution for the problem. |
| 13:22.18 | Samot | And, as I said, it's causing more issues than it's worth. |
| 13:22.27 | [TK]D-Fender | Is there actually a problem? |
| 13:22.34 | Samot | Well for them it is. |
| 13:22.50 | [TK]D-Fender | I though his was just a painful request that should ahve been the sme day he fistt came in |
| 13:22.57 | Samot | Their end users are here their upstreams recordings about no funds or their SIM card being blocked. |
| 13:23.14 | Samot | s/here/hearing/ |
| 13:23.49 | Samot | And the urgency for this was because one of their customers was going to switch providers because they kept getting the messages. |
| 13:24.07 | Samot | Jack17: That about right? |
| 13:26.00 | jack17 | nooo |
| 13:26.10 | Samot | Then what is this AMD solution for? |
| 13:26.18 | jack17 | i only want to pass call in street |
| 13:26.24 | jack17 | quite calls shouldnt passs |
| 13:26.34 | jack17 | while there is traffic sound |
| 13:26.38 | jack17 | pass call |
| 13:26.45 | Samot | That's how a CALL WORKS |
| 13:26.47 | jack17 | if calls are made from home or no voice detect drop it |
| 13:26.58 | Samot | Why do you need AMD? |
| 13:27.24 | Samot | Because now you are giving reasons and scenarios that your "partner" never gave. |
| 13:27.34 | Samot | In fact, quite the opposite. |
| 13:28.51 | Samot | Again why would I call a number that answers in pure silence? |
| 13:29.43 | *** join/#asterisk dokma (~vlatko@cpe-94-253-187-31.zg.cable.xnet.hr) |
| 13:30.30 | jack17 | same => n,GotoIf($[ "${AMDCAUSE}" ="MAXWORDLENGTH" ]?${EXTEN:1},1) |
| 13:30.48 | jack17 | maxwordlength is a variable in amd |
| 13:30.51 | jack17 | can i do like that |
| 13:30.55 | Samot | WHAT? |
| 13:31.02 | Samot | WHAT THE FUCK ARE YOU TRYING TO DO? |
| 13:32.07 | jack17 | I told you if silence stop call if word>1 detected pass call |
| 13:32.21 | jack17 | amd has 3 varaibles as a status |
| 13:32.25 | jack17 | human machine and not sure |
| 13:32.33 | Samot | HOW is this call being made? |
| 13:32.34 | jack17 | notsure will result in silent and in too many words detected |
| 13:32.47 | jack17 | made from my asterisk |
| 13:32.51 | Samot | By a person or is Asterisk generating a call? |
| 13:32.52 | jack17 | coming from a client |
| 13:32.55 | jack17 | person |
| 13:32.56 | Samot | OK |
| 13:33.10 | Samot | So why does the caller, making a normal call, need AMD? |
| 13:33.22 | Samot | Either they are going to get a person and talk.. |
| 13:33.31 | Samot | OR they get a voicemail and they leave a message. |
| 13:33.38 | Samot | That's IF the call is ANSWERED |
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| 13:33.50 | Samot | That's how people make phone calls. |
| 13:34.10 | Samot | Why does AMD need to be interjected? |
| 13:34.37 | jack17 | https://gist.github.com/anonymous/044527be0283eac9689e2aa4bf9f3c49 |
| 13:34.48 | jack17 | look at the 3 scenarios please to understand |
| 13:34.54 | Samot | Why does AMD need to be interjected? |
| 13:35.13 | jack17 | man i need to stop the silence calls |
| 13:35.15 | jack17 | i dont want it |
| 13:35.22 | Samot | What silent calls? |
| 13:35.41 | Samot | Again, as a user why would I be calling numbers that answer with silence? |
| 13:35.56 | Samot | As the USER I would just hang up. |
| 13:36.34 | Samot | Why are YOU interfering with a USERs call? |
| 13:39.48 | jack17 | 0007", "THIS IS MACHINE") in new stack |
| 13:39.48 | jack17 | <PROTECTED> |
| 13:40.10 | jack17 | i want to stop when INITIALSILENCE IS happening |
| 13:40.18 | Samot | ??? |
| 13:40.26 | Samot | You have no idea how calls work. |
| 13:40.36 | jack17 | same => n,GotoIf($[ "${AMDCAUSE}" ="INITIALSILENCE" ]?${EXTEN:1},1) |
| 13:40.43 | jack17 | can i use this ? |
| 13:41.00 | Samot | The Initial Silence is how long it takes from me picking up the handset and saying hello. |
| 13:41.06 | Samot | You need initial silence. |
| 13:41.25 | Samot | Machines and humans will have different lengths. |
| 13:41.51 | Samot | If you kill a call during the initial silence phase then you are just hanging up the call when it answers. |
| 13:42.32 | Samot | There is going to be at least 1 to 2 seconds of silence between the ANSWER and someone/thing speaking. |
| 13:44.31 | Samot | Still doesn't answer why you are interfering with your clients calls. |
| 13:44.50 | Samot | I wouldn't care if a client called a destination that played back 30 seconds of silence. |
| 13:45.00 | Samot | They're still paying me to listen to nothing for 30 seconds. |
| 13:45.38 | jack17 | man we have 3 scenario in amd |
| 13:45.41 | jack17 | Machine |
| 13:45.45 | jack17 | NOTSURE AND HUMAN |
| 13:45.51 | Samot | Right. |
| 13:46.00 | Samot | WHY are you INTERFERING with the CALL? |
| 13:46.02 | jack17 | when you call and be silent it will result as NOTSURE TOOLONG |
| 13:46.18 | jack17 | when you call and sing and dont stop talking it will result as notsure toolong |
| 13:46.33 | Samot | OK. I'm done. |
| 13:46.38 | jack17 | same results different scenarios |
| 13:46.53 | jack17 | im doing this because i want to |
| 13:47.02 | jack17 | i dont have to explain because you will not get my point |
| 13:47.05 | jack17 | im doing a special task |
| 13:47.19 | [TK]D-Fender | And we're not capable of understanding |
| 13:47.22 | jack17 | you are the expert but you shouldnt care why would i want do this for my clients |
| 13:47.32 | [TK]D-Fender | This from the guy spending weeks on a tiny piece of dialplan |
| 13:47.37 | Samot | You're not doing this for your clients. |
| 13:47.48 | Samot | You're interfering with their ability to make calls. |
| 13:47.52 | jack17 | hehe |
| 13:47.55 | jack17 | Samot |
| 13:48.07 | Samot | More importantly, you're interfering with their ANSWERED/CONNECTED calls. |
| 13:48.18 | [TK]D-Fender | <jack17> same => n,GotoIf($[ "${AMDCAUSE}" ="INITIALSILENCE" ]?${EXTEN:1},1) |
| 13:48.18 | [TK]D-Fender | <jack17> can i use this ? <- why are you even ASKING? |
| 13:48.21 | [TK]D-Fender | Does it WORK? |
| 13:48.27 | jack17 | i tried nnoo |
| 13:48.50 | [TK]D-Fender | Can you even handle the simple task of COMPARING a variable in asterisk dialplan? |
| 13:48.52 | jack17 | [TK]D-Fender apparently you cant use amdcause results with gotoif |
| 13:49.01 | [TK]D-Fender | Nonsense |
| 13:49.12 | [TK]D-Fender | YOU have no idea what you are doing or looking at |
| 13:49.18 | Samot | ^ That |
| 13:49.34 | [TK]D-Fender | PB your dialplan and a call |
| 13:49.36 | *** part/#asterisk anpi (~crr@unaffiliated/anpi) |
| 13:51.03 | Samot | I generally have about a 3-5% false positive with about 1800 calls using AMD. |
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| 13:59.42 | Samot | 10 minutes later........ |
| 14:02.07 | jack17 | https://gist.github.com/anonymous/0d57660eaccfb545c85a37c29c2b3d37 |
| 14:02.09 | jack17 | [TK]D-Fender |
| 14:02.28 | jack17 | 2 calls here |
| 14:02.33 | jack17 | the 1st on i dont want it to pass |
| 14:02.38 | jack17 | but the second call i want it |
| 14:03.46 | Samot | That's not even an entire call. |
| 14:04.04 | Samot | Need to see a FULL call from start to finish. |
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| 14:34.11 | [TK]D-Fender | Another half hour later....... |
| 14:37.53 | Samot | Yup. |
| 14:38.26 | Samot | Just a complete waste of time to do something that interferes with clients connected outgoing calls. |
| 14:39.45 | Samot | The reasoning that went from "we don't want upstream carrier recordings heard by end users, so if they happen we want to hang up the call" to "no man, we just want to hang up silent calls". |
| 14:40.02 | Samot | "silent" calls. |
| 14:40.19 | [TK]D-Fender | No carrier uses SIM cards to run a telco, especially ones that will cut them off and shove in-band messages. That's the sort of thing someone using CLONED SIMs and trying to abuse plans not meant for resale would do. |
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| 14:40.43 | [TK]D-Fender | But I don't even care about thatt |
| 14:41.39 | Samot | Just a debug call and dialplan.. |
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| 15:12.25 | [TK]D-Fender | And now over an hour |
| 15:12.34 | [TK]D-Fender | And we wonder why tthis has been going on for weeks |
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| 15:16.11 | Samot | No. We know why. |
| 15:16.14 | Samot | Let's not kid. |
| 15:17.07 | Samot | He's busy thinking up new names to use and how it will be associated to this. |
| 15:17.18 | Samot | He's figured out this alias is burned. |
| 15:20.33 | [TK]D-Fender | He's about to become a new "coworker"! |
| 15:22.23 | Kobaz | anyone familiar with MeetMe's denoiser option? |
| 15:22.29 | Kobaz | i'm going to try without it |
| 15:22.37 | Kobaz | i have massive complaints of call quality issues |
| 15:24.03 | Kobaz | well it says by default it turns it on |
| 15:24.17 | Kobaz | so, i'm actually adding the n, for no denoiser |
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| 15:35.25 | EnrgySmth | Wanting to do a little project for my 18mo old Grand Daughter who is living with us and separated from her Mom and Dad. She is obsessed with phones. Was hoping to setup some type of a phone system so she can listen to recordings from her Mom, Dad, other Grandparents, and Aunts, without any fear of her dialing emergency services or Kazakhstan. I got a Raspberry PI 3 model B, Cisco SPA112 and Cheap VTech Cordless Phone. |
| 15:35.27 | EnrgySmth | Obviously needs to be simple because she tends to push a lot of buttons. Anyone have any ideas or guidance on where to start or what will/wonât work? |
| 15:36.22 | [TK]D-Fender | depends how you want it to reactt to dials |
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| 15:37.15 | [TK]D-Fender | so if any button immediately plays a recording then that max 12 recordings. You'd set the ATA to match single-digit patterns and send immediately. |
| 15:37.39 | EnrgySmth | Oh nice that would be perfect |
| 15:38.11 | [TK]D-Fender | Or set it to dial "empty" immediately and then do everything as an IVR |
| 15:38.26 | [TK]D-Fender | So it goes off-hook and instantly auto-dials the server |
| 15:39.16 | EnrgySmth | I wonder if there is a way to get an IVR to play a random message |
| 15:39.24 | EnrgySmth | googling |
| 15:40.39 | [TK]D-Fender | of course |
| 15:40.47 | [TK]D-Fender | it's your dialplan, it does whatever you tell it to |
| 15:41.59 | EnrgySmth | [TK]D-Fender: thank you for your guidance, it is much appreciated |
| 15:42.14 | [TK]D-Fender | You're welcome |
| 15:42.57 | EnrgySmth | so if it was that any button immediately plays a recording, is that using the Announcements feature? |
| 15:43.41 | EnrgySmth | I might start with that, and then play with the IVR idea after |
| 15:43.56 | [TK]D-Fender | By that phrasing you're using FreePBX which isn't really supported here |
| 15:44.26 | [TK]D-Fender | when people come in here asking "can I X" it is assumed they are actually managing Asterisk themselves and answer are given based on that. |
| 15:44.56 | EnrgySmth | Ok, thank you, I understand |
| 15:45.47 | [TK]D-Fender | Answering based on FP gets trickier. You'd have to point that "extension" to a context that doesn't let them dial out, etc (like one created by a FP IVR) and then they'd be restricted tto those options |
| 15:46.20 | [TK]D-Fender | Where you could easily pick an annoucement for each digit, but it wouldn't be random |
| 15:46.29 | [TK]D-Fender | random requires custom dialplan |
| 15:48.43 | EnrgySmth | Just looking at http://www.raspberry-asterisk.org/ it says it comes with both Asterisk and FreePBX. Is it either/or or do they compliment each other? |
| 15:50.10 | [TK]D-Fender | FreePBX is a GUI interface that takes over the primary configuration of EVERYTHING for Astterisk |
| 15:50.33 | [TK]D-Fender | You don't tend to touch configs yoruself ever. FreePBX generates them 100% for you from scratch every time you apply its changes |
| 15:50.48 | EnrgySmth | Ah, I see |
| 15:51.10 | [TK]D-Fender | It is not like Webmin which parses configs to do its changes, but rather full regeneration |
| 15:51.27 | [TK]D-Fender | tthis lends to somewhat more limited ability to customize specific behaviours |
| 15:54.14 | EnrgySmth | Ok, will have to experiment tonight and get my feet wet. Thank you again [TK]D-Fender! |
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| 15:55.32 | [TK]D-Fender | Tip for you : create the extension and the IVR. In the extension set the context from being "from-internal" to say "ivr-#" where that is the # you see in the URL when editing the IVR you want to lock it to |
| 15:56.10 | [TK]D-Fender | it will prevent it from being able to have the same rights as your "regular" phones |
| 15:56.24 | EnrgySmth | outstanding |
| 15:57.01 | EnrgySmth | I will stop back when I have the thing together and in front of me. |
| 15:57.32 | [TK]D-Fender | If you wanted to learn the "100%" way you could also just set the context to one you name yourself and then code in its entirety in extensions_custom.conf |
| 15:57.51 | [TK]D-Fender | Which would allow you to do ... litterally whattever you wanted |
| 15:57.54 | EnrgySmth | ooo |
| 15:59.49 | [TK]D-Fender | Each ivr normally "includes" a context named [ivr-#-custom] which you can create in that conf file tto add JUSt the custom bits so you can learn in a hybrid way |
| 16:01.01 | EnrgySmth | Ok, I will read up on that |
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| 18:09.26 | igcewieling | heh, the cable company REALLY REALLY doesn't like taking services off. |
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| 18:17.03 | Kobaz | anyone have problems with meetme and the fact that denoiser is turned on by default? |
| 18:18.03 | igcewieling | people still use meetme? |
| 18:18.13 | Kobaz | confbridge doesn' |
| 18:18.20 | Kobaz | t yet have all the same options as meetme does |
| 18:19.07 | igcewieling | well, good luck, I guess. |
| 18:19.45 | Kobaz | well i turned on 'n' |
| 18:19.50 | Kobaz | which is no denoiser |
| 18:19.56 | Kobaz | and it seems to have fixed the quality issues |
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| 18:22.38 | igcewieling | you could try the non-meetme denoiser / agc |
| 18:23.04 | Kobaz | agc? |
| 18:23.44 | igcewieling | automatic gain control. there is a DENOISE and an AGC function if (Speex) is compiled in. |
| 18:24.37 | Samot | Uhm denoise=yes is a confbridge_app setting |
| 18:24.43 | Samot | I don't think it exists in meetme |
| 18:24.52 | Samot | At least not in any of the core sample configs. |
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| 19:01.15 | klow | Anyone have a programmatic way to test conference call quality/jitter etc? I want to stress test some VM configurations without going around to a bunch of desks to dial into the conf heh |
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| 19:01.41 | klow | I looked at "timing test" command in asterisk .. but Im not sure how well that will tell me how good a conf call will work. |
| 19:02.07 | klow | Im starting to wonder if our firewalls are the source of the conf call jitter and not the VM/clock since it seems to happen both on ESXi and KVM |
| 19:02.27 | klow | normal calls fine, get 3-4 ppl on a conf and it is just all kinds of skippy |
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| 22:00.39 | Samot | klow: where are all the users calling in from? Locally or via PSTN? |
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| 22:01.23 | klow | they are all just registered to the same freepbx instance, from various locations, and they dial the conf 4 digit as setup in the freepbx webUI. |
| 22:01.45 | klow | Im actually building a bare metal freepbx system right this second, going to test it on my LAN, then move it to datacenter, then behind the firewall |
| 22:01.47 | Samot | So they are remote users? |
| 22:02.13 | Samot | They are using bandwidth to be part of the call? |
| 22:02.14 | klow | ya. thats our whole thing is like clients are everywhere. mobile apps, desk phones at home offices, the whole nine |
| 22:02.33 | Samot | Are you recording the conf calls? |
| 22:02.35 | klow | its tough. but theres something pretty specifically wrong with conf calls for us |
| 22:02.41 | klow | no |
| 22:02.51 | Samot | How many other users on the system? |
| 22:02.55 | Samot | And calls? |
| 22:03.09 | Samot | Or it is just these people in the conf? |
| 22:03.23 | klow | we of course have some spotty non-conf calls, where you can hear audio cut out or whatnot. but this is different. its like a clock skipping issue, something with transcoding. a very regular sounding choppiness |
| 22:03.45 | Samot | Very well could be. |
| 22:03.47 | klow | these are all VMs, we have ESXi and KVM and it seems to happen on both |
| 22:03.50 | Samot | That was going to be next. |
| 22:03.54 | klow | they all provisioned well, and not busy |
| 22:03.59 | Samot | Are all the devices using the same codec? |
| 22:04.31 | Samot | 90% of my PBX systems are VMs. |
| 22:04.33 | klow | not necessarily. we have some g729 (mobil) and some ulaw . I think we have tested with everyone on ulaw before though |
| 22:04.44 | Samot | OK, that's your problem. |
| 22:04.55 | igcewieling | 0% of my PBX systems are in VMS. |
| 22:05.32 | klow | I agree it could be the problem. but its not CPU cycles . its clocking I think. |
| 22:05.42 | Samot | The only PBX systems I have that aren't VMs are on premise systems. |
| 22:05.45 | klow | I've watched the CPU when the problem is occuring and its doing very little even w/transcoding |
| 22:05.56 | Samot | klow: Test with all on ulaw. |
| 22:05.59 | Samot | See what happens. |
| 22:06.02 | Samot | Or all on g279. |
| 22:06.07 | Samot | Does the issue go away? |
| 22:06.13 | Samot | Then it's transcoding. |
| 22:06.16 | klow | Will test this for sure. |
| 22:06.24 | klow | Also, importantly .. we use SRTP across the board. |
| 22:06.32 | Samot | One guy on g279 and the rest on ulaw... |
| 22:06.34 | klow | We are a privacy company, not really a voip company. Just 1 thing we do |
| 22:06.43 | Samot | Have to have three channels transcoded |
| 22:06.48 | Samot | and then encrypted. |
| 22:07.15 | klow | Ya I agree that is a lot of steps. But we are talking about beefy blade servers, with like 4 people on a conf call |
| 22:07.22 | klow | with CPU 99% idle |
| 22:07.25 | klow | or whatever |
| 22:07.47 | Samot | But that doesn't mean the transcoding isn't the actual cause. |
| 22:08.01 | Samot | Compressing/decompressing into the best format. |
| 22:08.15 | Samot | That alone, regardless of CPU can cause issues. |
| 22:08.42 | klow | I have been thinking about trying to move *everything* to OPUS , since OPUS transcoding is now supported "natively" in Asterisk 14 |
| 22:08.54 | klow | Im using it a bit already |
| 22:09.08 | klow | but then I read that using VBR codecs with SRTP is a possibly security issue too |
| 22:09.12 | klow | its always something heh |
| 22:09.33 | Samot | I try to avoid transcoding as much as possible. |
| 22:14.31 | *** join/#asterisk miralin1 (~Thunderbi@194.8.128.48) |
| 22:15.54 | drmessano | Literally haven't seen a timing VM problem since 1.4 |
| 22:16.12 | Samot | ^^^^^ |
| 22:16.19 | drmessano | Which was less about Asterisk and more about Kernel 2.6, I think |
| 22:16.21 | Samot | Well that's not true. |
| 22:16.49 | Samot | But then again, it was like 2,500 calls at onces all doing audio playback and AMD. |
| 22:17.41 | Samot | Plus it was behind NAT on AWS. |
| 22:17.48 | Samot | So it had a lot going against it. |
| 22:17.53 | drmessano | Literally 99.99% of the time the argument devolves to "I remember in 2004 I needed Zaptel installed so I could have the dummy module or Asterisk wouldn't the anything" |
| 22:18.07 | drmessano | Which even for 2004 was horribly misguided |
| 22:19.07 | Samot | Try this.. |
| 22:19.22 | Samot | g729 -> g711u -> S7 switch |
| 22:19.33 | drmessano | lol |
| 22:19.44 | Samot | I shit you not. |
| 22:19.58 | Samot | That was the Telecom's first SIP rollout plan. |
| 22:20.06 | Samot | g279 because it was on T1's.. |
| 22:20.22 | Samot | But no IP peering to PSTN at that point all TDM |
| 22:20.28 | Samot | So out that GD S7. |
| 22:35.29 | aandrew | hm, perhaps you'd know samot. doesn't asterisk avoid transcoding unless absolutely required? |
| 22:35.52 | Samot | Yes. |
| 22:36.18 | Samot | But if device A is g729 and device B is ulaw... |
| 22:36.22 | Samot | Transcoding. |
| 22:36.30 | aandrew | SIP/phone forces ulaw. SIP/provider supports g729,gsm,ulaw (in that order in sip.conf). a call from SIP/phone out through SIP/provider shows the simple bridge as ulaw-slin-g729 even though the provider could do ulaw |
| 22:36.45 | aandrew | confirmed by forcing SIP/provider sip.conf entry to ulaw only |
| 22:36.54 | Samot | Right. |
| 22:37.12 | aandrew | so is the preference order overriding asterisk's "avoid transcode unless necessary" ? |
| 22:37.26 | Samot | He's talking about "local" devices. |
| 22:37.33 | Samot | On a conference call |
| 22:37.44 | aandrew | a conf call always mixes in slin though |
| 22:37.44 | Samot | With a mix of g729 channels and ulaw channels. |
| 22:38.08 | aandrew | to my knowledge you can't mix coded streams |
| 22:39.01 | Samot | If my phone ig g729 only... |
| 22:39.13 | Samot | Asterisk and phone channel is g729. |
| 22:39.36 | aandrew | Samot: but if you have 3 g729 calls in a conf you're still gonna be transcoding to do the mix |
| 22:39.36 | Samot | So Asterisk has to send the audio in a format best for g729. |
| 22:39.53 | aandrew | you can bridge without transcoding but you can't mix audio without transcoding to slin |
| 22:42.36 | Samot | If one channel is ulaw (PBX <-> Provider) and the other channel is g729 (PBX <-> Phone)... |
| 22:42.48 | Samot | That audio has to be compressed for g729. |
| 22:42.55 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 22:46.17 | aandrew | sure |
| 22:47.30 | aandrew | but (for my issue above) my question is if phone is ulaw only and provdier can do g729,gsm,ulaw, why is asterisk negotiating g729 when ulaw is available and the general idea is "dont' transcode unless necessary" ? The only thing I can think of is that the sip.conf order of the codecs is overriding the idea of not transcoding but that doesn't seem to make sense |
| 22:48.00 | drmessano | It doesn't make sense |
| 22:48.02 | Samot | Except this has nothing to do with a provider. |
| 22:48.09 | aandrew | for a conf though it doesn't matter if *all* participants are g729, every stream is being transcoded to slin, combined, then the combined stream transcoded back out to g729 for each participant |
| 22:48.22 | aandrew | Samot: I agree |
| 22:48.30 | Samot | So don't relate it. |
| 22:48.32 | aandrew | it could be SIP/phone1 and SIP/phone2 with the same sip.conf settings mentioned above |
| 22:49.02 | aandrew | I wasn't relating it to a provider at all, just two sip accounts |
| 22:49.22 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 22:49.25 | Samot | aandrew: He's relating choppy audio in a conference call to CPU timing issues on a VM. |
| 22:49.47 | Samot | He said that users are REMOTE and various devices are using various codecs. |
| 22:49.48 | aandrew | Samot: I think we are having two similar but different conversations |
| 22:49.53 | Samot | We are. |
| 22:50.09 | aandrew | for klow, if he's conferencing, he's transcoding. no way around it |
| 22:50.15 | Samot | RIGHT |
| 22:50.18 | Samot | WHAT I SAID |
| 22:50.19 | klow | heh ;) |
| 22:50.21 | Samot | THE ENTIRE TIME |
| 22:50.21 | drmessano | If you want to know how something is being negotiated you could, I don't know, look at the debug. |
| 22:50.45 | aandrew | for voicemail though... unless voicemail is set up to save in mp3 or whatever only, it should not have to transcode. chances are it's not being saved in g729 format though |
| 22:51.02 | Samot | Voicemail, by default, is wav |
| 22:51.09 | aandrew | drmessano: sure, and that's the next step. I was just asking if there was something I was overlooking before that |
| 22:51.20 | drmessano | Im the only person with G729 voicemail |
| 22:51.24 | aandrew | heh |
| 22:51.28 | aandrew | probably |
| 22:51.54 | Samot | I also believe klow confirmed that all the testing that passed was done with all ulaw based devices.. |
| 22:52.03 | aandrew | yep mine's wav (just checked) |
| 22:52.06 | Samot | So that does point to a codec transcoding issue. |
| 22:52.23 | aandrew | right, ulaw/slin is almost a noop in cpu time |
| 22:52.28 | Samot | And he's going to confirm. |
| 22:52.45 | aandrew | I didn't see in the history, are his devices all SIP or are there TDM devices in the mix? |
| 22:54.10 | Samot | No. |
| 22:54.24 | Samot | You took two conversions and drew a new one out of it. |
| 22:54.55 | Samot | drmessano and I started on a conversation about transcoding nightmares. |
| 22:55.50 | Samot | Segues... |
| 22:55.57 | Samot | They're tricky. |
| 22:56.04 | aandrew | right, it was a segue |
| 22:56.06 | aandrew | < aandrew> hm, perhaps you'd know samot. doesn't asterisk avoid transcoding unless absolutely required? |
| 22:56.28 | Samot | 5:35:52 PMÂ <Samot>Â Yes. |
| 22:56.38 | Samot | Asked / Answered. |
| 22:56.40 | aandrew | yes, and then I went into more detail about my question |
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| 23:05.11 | eph1610 | hello |
| 23:08.34 | eph1610 | I'm trying to get Cisco SPA phones working with asterisk. All seems to work except blind transfers. I am getting SIP 603 declined errors back on all REFERs. Any ideas what might be wrong. Polycom phones are able to blind transfer on the same system. |
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