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| 03:15.12 | jjmil03 | hello |
| 03:16.03 | jjmil03 | is anyone on? I had a few questions if someone doesn't mind helping me out for a bit |
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| 03:17.32 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb26808c37.vs.shawcable.net) |
| 03:17.41 | jjmil03 | hello |
| 03:21.09 | [[thufir]] | channels: http://pastebin.com/qK3GKgPz and contexts: http://pastebin.com/eekQtHdb dialing in works fine, it's dialing out that's troublesome. I don't know where the call breaks down. I want to, I think, use the BABY context variable in the dialplan, but I'm not sure how. It seems circular, that the context uses the dialplan, but then it goes back. Obviously not so, just that I'm confuses. |
| 03:23.11 | [TK]D-Fender | [[thufir]], Show us an actual failure. |
| 03:23.25 | [TK]D-Fender | [[thufir]], Otherwise we have no idea if you are dialing something pointless |
| 03:25.14 | [[thufir]] | [TK]D-Fender: that might be, I wonder if it's not something about the prefix 9 versus 1. Asterisk plays either demo-instruct.gsm or invalid.gsm depending on how it's dialed. I'll recreate a failure and post the messages log? That's where the .gsm files show. |
| 03:25.36 | [TK]D-Fender | [[thufir]], Show us an actual failure |
| 03:25.48 | [[thufir]] | ok, be a few. thanks. |
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| 03:30.26 | [[thufir]] | hmm. it played invalid.gsm but there's nothing recent in /var/log/messages about that or an attempted dial... |
| 03:31.58 | [TK]D-Fender | <[TK]D-Fender> [[thufir]], Show us an actual failure |
| 03:35.30 | [TK]D-Fender | jjmil03, .. |
| 03:35.34 | [TK]D-Fender | ~ask |
| 03:35.42 | 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. |
| 03:35.46 | [TK]D-Fender | ^^ |
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| 03:39.14 | phix | [TK]D-Fender: Faulty ADSL filter or cable |
| 03:39.22 | phix | thnx for your assist yesterday |
| 03:39.51 | [TK]D-Fender | phix, I saw it before we left |
| 03:40.08 | phix | ah ok great |
| 03:40.10 | phix | gg |
| 03:41.38 | [[thufir]] | asterisk log of failed dial out: http://pastebin.com/TcuNzTRJ |
| 03:42.32 | [[thufir]] | [TK]D-Fender: line 177 it's playing demo-congrats.gsm rather than dialing the number |
| 03:44.04 | [TK]D-Fender | [Apr 8 23:38:52] VERBOSE[3279] pbx.c: [Apr 8 23:38:52] == Starting Local/8600051@default-00000000;1 at default,917782934001,1 failed so falling back to exten 's' |
| 03:44.12 | [TK]D-Fender | Your sending calls to places that don't exist |
| 03:44.15 | [TK]D-Fender | you're* |
| 03:47.55 | [[thufir]] | [TK]D-Fender: i follow, but I don't. the log shows, the part you quoted, a "non" number? that's like dialing 00000 on a phone? |
| 03:48.30 | [TK]D-Fender | No. |
| 03:48.46 | [TK]D-Fender | You are originating a channel... and sending it into the dialplan ... to a place that DOES NOT EXIST |
| 03:48.57 | [TK]D-Fender | <PROTECTED> |
| 03:51.39 | [[thufir]] | shoot. I gotta go in ten. I see in the book dial(${RUSSELL}) you're talking about the syntax to use the dial app? and, I'm passing in a number which doesn't exist as a context? |
| 03:52.11 | phix | [[thufir]]: Can I dial that number? |
| 03:52.37 | [[thufir]] | phix: ok |
| 03:54.21 | [TK]D-Fender | [[thufir]], Doesn't matter what you see in some book |
| 03:54.26 | [TK]D-Fender | Look at what you are DOING right ther |
| 03:54.34 | [TK]D-Fender | You are sending a call to a place in your dialplan. |
| 03:54.38 | [TK]D-Fender | It does not EXIST |
| 03:57.17 | phix | :D |
| 03:57.27 | phix | Can I try and sell you something? |
| 03:57.31 | [[thufir]] | pls do |
| 03:57.42 | [[thufir]] | phix: no, i thought said tell |
| 03:58.17 | [[thufir]] | I'm going to read a bit. thanks for the help. bye. |
| 04:13.23 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 04:13.26 | linocisco | hi all |
| 04:19.02 | phix | hehe |
| 04:19.02 | phix | :P |
| 04:19.04 | phix | hai linocisco!@ |
| 04:19.25 | linocisco | phix, hi. why funny? |
| 04:22.33 | WIMPy | phix! |
| 04:25.18 | linocisco | WIMPy, [TK]D-Fender , you two are ever alive on asterisk channel |
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| 08:11.22 | phix | linocisco: i was laughing at what i said to [[thufir]] just before you came on |
| 08:11.57 | phix | WIMPy: wooooooo! |
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| 08:50.47 | Eric-K | I'm testing Asterisk 13 with PJSIP and have a question regarding T.38 fax. My provider supports T.38 so I enabled it on the endpoint in pjsip.conf and would like to send a file (tiff from the astdatadir) to the provider but can't seem to figure out the right extensions dialplan for sendfax. |
| 08:50.51 | Eric-K | Does anyone have an idea? |
| 08:51.38 | Eric-K | I can SendFax(${ASTDATADIR}/send.tiff) but how do I make it connect to the SIP provider? |
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| 08:53.17 | wdoekes | Eric-K: you need 2 legs: see 'channel originate' |
| 08:53.18 | wdoekes | Usage1: channel originate <tech/data> application <appname> [appdata] |
| 08:53.27 | Eric-K | check |
| 08:53.44 | wdoekes | tech/data would be SIP/provider/foo |
| 08:53.52 | wdoekes | appname would be SendFax |
| 08:54.22 | wdoekes | (although you probably need a bit more, so you'd use the 'exten' instead of 'application' mode) |
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| 08:54.35 | wdoekes | (where you Answer(), Wait() and then SendFax()) |
| 08:55.41 | Eric-K | Let's give this a try. |
| 09:10.08 | Eric-K | Can't get it working so I am trying to setup a simple call to start with wdoekes. |
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| 09:10.17 | Eric-K | channel originate PJSIP/Motto/NUMBERHERE application Milliwatt |
| 09:10.43 | Eric-K | Could not create dialog to endpoint '...' as URI '...' is not valid |
| 09:13.08 | Eric-K | ah nvm, found it lol :P |
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| 09:18.26 | stefan27 | Does pjprojects have an irc channel like this |
| 09:18.44 | stefan27 | Couldnt find with a simple google search at least |
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| 09:30.10 | Eric-K | just fyi wdoekes, got it working like this: channel originate PJSIP/FAXNUMBER@PROVIDER application SendFax /etc/asterisk/Va.tiff |
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| 09:40.22 | Zetten | Is there any way to control the PJSIP caller display name for unknown incoming callers? Right now we get a UUID where the incoming number is blocked and it's confusing the poor users |
| 09:40.36 | Zetten | Ideally it would be just a generic "Unknown number" or something like that |
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| 10:00.30 | wdoekes | Eric-K: good |
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| 10:32.25 | ectospasm | In queues.conf, does a timeout of 0 mean that callers will wait indefinitely in the queue? I couldn't find documentation that explicitly states this. |
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| 13:58.35 | qakhan | Hi All, is there any way while call in queue and caller press 1 then call come out from queue go to voice mail. |
| 14:01.18 | WIMPy | Sure |
| 14:01.44 | [TK]D-Fender | qakhan: Read the queues.conf.sample |
| 14:01.44 | WIMPy | You just have to configure a context with possible extensions. |
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| 14:06.07 | qakhan | ok |
| 14:06.13 | qakhan | thanks guys |
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| 14:12.54 | emk | hi all I have an asterisk pbx setup (and it is routing calls properly to many windows machiens running qutecom)... Now which opensource SIP client can I download and use on Linux? |
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| 14:16.15 | [TK]D-Fender | ~softphone |
| 14:16.23 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 14:16.36 | trurl | Can someone suggest a SIP-Client for iOS? I've tried Zoiper Premium, but i'm not receiving any audio (without any errors) when calling the "hello-world" example. |
| 14:16.50 | *** join/#asterisk kharwell (kharwell@nat/digium/x-uklazsedjigsixae) |
| 14:16.57 | [TK]D-Fender | trurl: Your client isn't to blame |
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| 14:18.26 | trurl | [TK]D-Fender: i guess it's a codec issue? im not there yet ;) |
| 14:18.43 | [TK]D-Fender | trurl: Nor is it a codec issue. |
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| 14:19.05 | [TK]D-Fender | trurl: it is either a networking, or SDP issue |
| 14:19.39 | trurl | what is it? the "hello-world example" works perfectly fine with a desktop phone |
| 14:20.08 | [TK]D-Fender | "desktop phone" say absolutely nothing of the networking circumstances involved between either or how anything is configured |
| 14:20.24 | [TK]D-Fender | trurl: Do you think they published software whose audio is simply completely broken? |
| 14:20.38 | [TK]D-Fender | trurl: Go look at your call. |
| 14:21.00 | [TK]D-Fender | "sip set debubg on" <--- |
| 14:21.04 | [TK]D-Fender | "sip set debug on" <--- |
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| 14:59.58 | kannan | how can we prevent logging for a few specific priorities oly when logger is set to 'full' |
| 15:00.25 | WIMPy | Use the Log application. |
| 15:01.03 | kannan | to keep track of dtmf sequences in ivr |
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| 15:02.19 | [TK]D-Fender | How does "prevent logging" support the idea of "keep track of"? |
| 15:02.28 | [TK]D-Fender | Tack .. DON'T TRACK! |
| 15:02.31 | [TK]D-Fender | Track* |
| 15:02.37 | kannan | sorry i misread WIMPy |
| 15:03.37 | kannan | Log(level, message), the next priority should not be in the log, and then resume 'full' log |
| 15:04.18 | kannan | if we use Log application , then does it set the level for all subsequent priorities? |
| 15:04.27 | [TK]D-Fender | no |
| 15:04.31 | kannan | the core show application Log is not so clear |
| 15:04.47 | [TK]D-Fender | It overrides the global level |
| 15:05.05 | [TK]D-Fender | Never assume a call to an app like that has a lingering effect |
| 15:05.10 | [TK]D-Fender | It is there for itself. |
| 15:05.25 | kannan | if i have an input of user code and then password, i dont want to log password priority, can that be done? |
| 15:05.56 | [TK]D-Fender | This doesn't change the active logging level.... |
| 15:06.49 | WIMPy | sees no way to do that |
| 15:07.25 | [TK]D-Fender | Perhaps issuing a CLI command the change it? |
| 15:07.48 | kannan | if i want to call a system application and use awk, can i put it in the h context, will the /var/log/asterisk/full be already written, that may work i think |
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| 15:08.29 | kannan | but with a CLI it cant change the log level for a particular channel, or can it? |
| 15:08.44 | [TK]D-Fender | No such thing |
| 15:08.51 | [TK]D-Fender | overall level is overall level |
| 15:08.55 | kannan | ok. |
| 15:09.13 | [TK]D-Fender | Which means they would be fallout of simultaneous calls |
| 15:09.18 | [TK]D-Fender | for* |
| 15:09.52 | kannan | can we run a shell command with system just after the priority , the log lines will already be written i think, if i can issue a wait-for-a-few seconds and awk or sed with a regex replace todelete that line |
| 15:10.04 | kannan | i will try that i think |
| 15:11.06 | kannan | or simply schedule a per minute shell script to delete that lines |
| 15:11.27 | kannan | just wanted to check if thee was any other way, thank you |
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| 15:47.10 | eppigy | Hello |
| 15:47.13 | eppigy | I am Dave |
| 16:02.34 | jzu_ | well this is funny - I got chan_dongle halfway working.. |
| 16:02.51 | jzu_ | I can dial through the Dongle but every number comes out as "The number is not in use, please check the number" :-D |
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| 16:15.26 | avb | guys, how to force reread realtime queue members? |
| 16:15.56 | avb | from ami or console |
| 16:16.04 | avb | except the core restart now :) |
| 16:19.26 | *** join/#asterisk samdaman (~swalker@bc104242.bendcable.com) |
| 16:24.14 | avb | nevermind, i just fixed it myself :) |
| 16:24.29 | avb | its appeared that uniqueid should be really unique on each change |
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| 16:37.06 | jzu_ | oh yes, I just had non-working dialplan on Asterisk |
| 16:37.09 | jzu_ | now it's all rightie |
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| 17:49.17 | samdaman | I'm wanting to fax using * 1.6 and have no idea where to start does anybody have a good tutorial or advice? |
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| 17:58.05 | avb | samdaman: rxfax/txfax works pretty good |
| 17:58.09 | avb | standart ones |
| 17:58.20 | avb | if you want something better hylafax is good |
| 17:58.43 | avb | samdaman: http://www.voip-info.org/wiki/view/Asterisk+fax |
| 17:59.08 | avb | there is also a better paid module from digium www.digium.com/sites/digium/files/fax-for-asterisk-manual.pdf |
| 17:59.36 | avb | you can create nice fax2email and email2fax with this |
| 17:59.38 | samdaman | how about using an ata connected directly to the fax .. no fax server |
| 18:13.04 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:16.58 | [TK]D-Fender | samdaman: Make sure you are using a provider that supports T.38. Or else |
| 18:18.11 | samdaman | my provider says it supports T.38 |
| 18:18.35 | samdaman | how do I enable T.38 on my end? |
| 18:19.06 | [TK]D-Fender | Read the sample config and any of the hundreds of guides out there for this... and buy an ATA that also supports it |
| 18:19.09 | *** join/#asterisk igcewieling (~ewieling@ip98-170-196-157.pn.at.cox.net) |
| 18:20.04 | samdaman | is T.38 built into 1.6 by default? |
| 18:21.01 | *** join/#asterisk theron_ (~theron@199.201.64.131) |
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| 18:22.33 | igcewieling | I'm looking for anyone who has successfully done T.38 passthrough from Carrier (Level 3) -> Asterisk (11.recent) -> Adtran Netvanta CPE (w/T.38)? I'm starting to think this setup is like unicorns or an honest politicions -- nice to think about but doesn't exist. I have this working fine and do not need help with it: Carrier (Level 3) -> Asterisk 11.x and spandsp/rxfax. |
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| 18:25.13 | igcewieling | samdaman: 1.6 isn't supported anymore and in fact is so old it doesn't even get bug fixes. |
| 18:25.30 | igcewieling | http://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
| 18:26.23 | samdaman | yes I know this is a pbx from fonality (not my choice) so I cannot upgrade |
| 18:27.28 | samdaman | not without breaking their hud anyway |
| 18:30.15 | samdaman | I would love to switch to something like freepbx and use isymphony for an operator panel but they paid for this so I'm stuck .. previous IT guy sold them on fonality |
| 18:32.44 | igcewieling | Did they also create an Internet Explorer only Intranet site? 9-| |
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| 18:35.51 | samdaman | I am required to use their web gui to make changes, custom changes made directly to configs get reverted every so often |
| 18:36.52 | samdaman | its not even a local web gui, if I'm not connected to the internet I can't change anything |
| 18:36.58 | samdaman | I hate it alot |
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| 18:49.37 | Eric-K | T.38 is driving me insane. I am able to connect with the t38fax.com service by using the SendFax application in my dialplan. I see SIP signalling, I see my Asterisk accepting the re-invite and I see an actual T.38 stream. I notice tho that my Asterisk only sends a couple of T.38 packets and that's it. The server at t38fax.com sends a T.38 stream but I'm not responding with a stream. Anyone a clue on how to troubleshoot this? Asterisk 13 and PJSIP instead |
| 18:49.39 | Eric-K | <PROTECTED> |
| 18:51.39 | [TK]D-Fender | Eric-K: Everything I've seen says * does not support T.38 Origination at all. |
| 18:51.58 | Eric-K | Hm, that explains a lot. |
| 18:52.21 | *** join/#asterisk theron (~theron@199.201.64.131) |
| 18:52.22 | Eric-K | It's only a gateway is what you're saying? |
| 18:52.53 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb26808c37.vs.shawcable.net) |
| 18:55.04 | [TK]D-Fender | Gateway allows to terminate T.38 IIRC |
| 18:55.53 | Eric-K | I think I'll find some T.38 softclient and use Asterisk as a gateway to pass on the T.38 stream to the t38fax service. |
| 18:56.09 | Eric-K | Asterisk 13 should be able to do that according to the wiki. |
| 18:57.41 | [TK]D-Fender | That's straight pass-through which should work. |
| 19:04.58 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 19:05.54 | bkruse | Hello my friends - when using ARI and the GET /ari/sounds/blah - is that real-time ? |
| 19:06.34 | bkruse | I can see the file in /var/lib/asterisk/sounds/en/blah.ulaw, and I can actually play it using Playback(blah), but when I do the ARI call, it says file not found, and from the command line, it says message not found |
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| 19:12.49 | Eric-K | [TK]D-Fender I just spoke to the good people at t38fax and they confirm the same. |
| 19:13.19 | [TK]D-Fender | Eric-K: One option = T38Modem + Hylafax |
| 19:13.47 | [[thufir]] | how did the asterisk language evolve? what language does it most resemble? |
| 19:14.00 | Eric-K | True, but in the end my Asterisk is going to passthrough only anyway. My testing setup was simply wrong :) |
| 19:17.01 | file | bkruse, it's not real time - it indexes at startup or when told to at the CLI |
| 19:17.53 | [TK]D-Fender | [[thufir]]: What "Asterisk Language"? Depending ona point of view there are almost a DOZEN. |
| 19:18.33 | bkruse | file: Gotcha, so I cannot use that to determine if an audio file has been written and is available. I also imagine that I shouldn't tell Asterisk to update the index all the time |
| 19:18.58 | file | it re-reads the entire directory, and also the file that contains the description for files |
| 19:19.00 | [[thufir]] | <PROTECTED> |
| 19:19.02 | file | it's not a light weight operation |
| 19:19.24 | [TK]D-Fender | [[thufir]]: What "language"? You are not bbeing SPECIFIC about your topic. |
| 19:20.32 | [[thufir]] | the syntax [general] context=trunkinbound (on two lines) is the language I refer to. |
| 19:20.44 | [TK]D-Fender | [[thufir]]: extensions.conf <----- |
| 19:21.11 | [TK]D-Fender | [[thufir]]: That is like the worst form of BASIC combbined with Assembler |
| 19:22.06 | [TK]D-Fender | [[thufir]]: And there are SEVERAL other formats possible. AEL and LUA directly. Then there is the concept of using AGI to do your dirty-work which will looks like whatever language you are doing your AGI in (except for the minimal sommand syntax) |
| 19:22.08 | igcewieling | Eric-K: I've never gotten T.38 to pass through Asterisk. |
| 19:22.34 | igcewieling | T.38 is the one place we can't use Asterisk. |
| 19:23.31 | [[thufir]] | hmm. in the guide to asterisk, they explain the syntax to a language, but, to my knowledge, never name the language. it's not BASIC, nor is it Assember, at least to my knowledge. It's not LUA, either. And, since AEL is new, it's not AEL. I don't know the name of the language. What's the primary language generally used for writing sip.conf and extensions.conf in tutorials? |
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| 19:25.04 | igcewieling | [[thufir]]: IT is a home grown languate |
| 19:25.19 | igcewieling | It is a homegrown language |
| 19:26.15 | [[thufir]] | what I mean, is, it must have a primary influence? I come from Java, and the syntax is strange. It would be helpful to understand the syntax of this language if I knew its origins and the name of this homegrown language. |
| 19:26.37 | [TK]D-Fender | [[thufir]]: Home-grown means there is NO origin |
| 19:26.49 | [TK]D-Fender | The INVEENTED it. Not lik or based on anything else |
| 19:27.34 | [[thufir]] | ok. does it have a formal name, or just the asterisk language? |
| 19:28.21 | [TK]D-Fender | "Asterisk dialplan" |
| 19:28.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:28.43 | [TK]D-Fender | maybe "extensions.conf dialplan" |
| 19:29.49 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 19:30.03 | [[thufir]] | thanks. |
| 19:30.25 | *** join/#asterisk superscrat (asanders@nat/digium/x-tvreuhwogxcwausy) |
| 19:31.15 | [TK]D-Fender | [[thufir]]: It has no origin. Does that help you understand the syntax to know that? |
| 19:32.26 | [[thufir]] | unfortunately, not. I was hoping that if were, for example, influenced by perl, that reading a perl book would help. |
| 19:32.51 | [TK]D-Fender | READ THE ASTERISK BOOK YOU WERE LINKED 2 OR 3 DOZEN TIMES |
| 19:33.01 | [TK]D-Fender | It explains how it works |
| 19:33.03 | bkruse | [TK]D-Fender++ |
| 19:35.17 | [[thufir]] | yes, I have the paper version. the chapters I'm looking at are focused on inbound calling, which is the focus of the book as a whole. |
| 19:35.39 | trurl | [TK]D-Fender: the problem (regarding the no-audio issue with zoiper on iOS) was pebcak. audio out was routed to bt-headphones. so there _was_ no audio out on the phone but also no error ;) |
| 19:36.03 | [TK]D-Fender | [[thufir]]: All calling is inbound. |
| 19:36.22 | [TK]D-Fender | [[thufir]]: DIAL is an application to have * call OUT. |
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| 19:40.01 | eppigy | [TK]D-Fender: you have been doing this for a long time man |
| 19:40.18 | eppigy | are you affiliated with Digi in some way? |
| 19:40.29 | [TK]D-Fender | is decripit |
| 19:40.29 | eppigy | or some other company? |
| 19:40.39 | [TK]D-Fender | nothing in the tech sector at all |
| 19:40.45 | eppigy | haha oh cool |
| 19:40.54 | eppigy | that is interesting |
| 19:41.02 | [TK]D-Fender | I work in the PLUMBING industry |
| 19:41.21 | [TK]D-Fender | That's the series of tubes that DON'T carry porn. |
| 19:41.29 | [TK]D-Fender | (normally) |
| 19:41.35 | eppigy | haha |
| 19:41.54 | eppigy | I used to do commercial plumbing construction a long time ago |
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| 19:55.14 | Cuzner | eppigy: I've asked him the same questions |
| 19:55.46 | Eric-K | igcewieling i can't get it to work either :( |
| 19:56.05 | [TK]D-Fender | I'm just an IT manager who can see the forest for the trees... |
| 19:58.12 | robmal | So, trunks. |
| 19:58.16 | Cuzner | IT manager at a plumbing company eh? |
| 19:58.17 | robmal | ;-) |
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| 20:03.23 | overyander | anyone up to helping me figure out a t.38 fax issue? |
| 20:03.56 | Chainsaw | overyander: What Asterisk version and what endpoints please? |
| 20:03.57 | samdaman | fax seems to be the topic of the day |
| 20:03.58 | Eric-K | lol, you're not the only one with a t.38 issue :P |
| 20:04.04 | Chainsaw | samdaman: I know, I had to check it wasn't the other guy. |
| 20:05.10 | Eric-K | T.38 passthrough does not seem to work either. I have the same problem between Asterisk 13 and the t38fax service. |
| 20:07.59 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 20:08.27 | Chainsaw | casts a critical look at Eric-K and overyander |
| 20:08.39 | Chainsaw | Has anyone ever seen them at a party together? |
| 20:08.46 | Eric-K | lol |
| 20:08.52 | Eric-K | where you from overyander |
| 20:09.59 | *** join/#asterisk djgerm (~Adium@c-73-170-173-171.hsd1.ca.comcast.net) |
| 20:10.54 | [[thufir]] | how is Dial(SIP/${EXTEN}@myprovider) distinct from Dial(${BABY}/${EXTEN}) ? I want to invoke the Dial application and send it the BABY parameter, which is a global string, BABY = SIP/babytel_out? That BABY is a global string which equates to a context confuses me. log of call being trapped, it never goes to the intended carrier: http://pastebin.com/MskMcUDi contexts: http://pastebin.com/jLwkUdQ2 channels http://pasteb |
| 20:11.40 | [TK]D-Fender | [[thufir]]: Depends what the VARIABLES parse out to be |
| 20:12.32 | [TK]D-Fender | [[thufir]]: and "myprovider" .... had better have a device entry in sip.conf ... or a host entry to resolve an IP. |
| 20:13.31 | overyander | Chainsaw, using * ver 11.2.1 not using endpoints, trying to get fax to e-mail working. |
| 20:13.32 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:13.36 | [TK]D-Fender | [[thufir]]: [[thufir]] Variables are dumb text. Theye don't equate to anything. bby the time you're done what matters is how those evaluate, and then how the APPLICATION treats what it is being passed |
| 20:13.45 | overyander | Eric-K, Arkansas |
| 20:13.45 | [[thufir]] | it would be babytel_out which has an entry in sip.conf for myprovider http://pastebin.com/VN6gP006 |
| 20:13.51 | Eric-K | <- netherlands |
| 20:13.58 | Chainsaw | overyander: Then you still are using endpoints. Asterisk does not have an SMTP plugin, so you are using a fax plugin of some sort. |
| 20:14.07 | Chainsaw | overyander: And what is the other end, anyhow? |
| 20:14.21 | [TK]D-Fender | [[thufir]]it would be babytel_out which has an entry in sip.conf for myprovider http://pastebin.com/VN6gP006 <- there is no [myprovider] in there |
| 20:14.31 | [TK]D-Fender | [[thufir]]how is Dial(SIP/${EXTEN}@myprovider) <- so this will fail |
| 20:14.59 | overyander | Chainsaw, right now i'm trying to recieve faxes consistantly. 9 out of 10 fail with session time-out error. |
| 20:15.01 | [TK]D-Fender | You also shouldn't use that syntax at all when referring to dialing out SIP peers you have defined |
| 20:15.27 | overyander | we're using a sip provider for termination, no PRI connected locally |
| 20:15.47 | [[thufir]] | Dial(SIP/${EXTEN}@babytel_out) is what I specifically mean. What syntax should I use when dialing out via SIP peers? |
| 20:16.32 | [TK]D-Fender | [[thufir]]: Dial(SIP/peername/numbertodial,timeout,options) |
| 20:16.38 | Chainsaw | overyander: T38 over SIP, dangerous territory. But very well, what codec is being negotiated prior to T38? |
| 20:17.47 | Cuzner | [04:03pm] <samdaman> fax seems to be the topic of the day - WHAT YEAR IS THIS?!? |
| 20:17.59 | overyander | Chainsaw, g711u |
| 20:18.17 | overyander | i have a filtered pcap of a working and failed fax if you'd like to see |
| 20:18.58 | Chainsaw | overyander: I'm afraid I'm on train WiFi with (diverse) GPRS, and can practically see the individual bits flying through the air. |
| 20:19.17 | overyander | lol |
| 20:19.20 | Chainsaw | overyander: Anything requiring more than a byte a second is in trouble. |
| 20:19.53 | [[thufir]] | [TK]D-Fender: how do I pass numbertodial as a variable? Dial(SIP/${EXTEN}@babytel_out) will dial out and...pass.. EXTEN ? |
| 20:20.55 | [TK]D-Fender | look at what I just gave you |
| 20:20.59 | *** join/#asterisk djgerm (~Adium@204.57.64.106) |
| 20:21.16 | samdaman | Cuzner, I know seems stupid to keep using this old method sure wish people would switch over to this new email thing I keep hearing about |
| 20:21.26 | [TK]D-Fender | [[thufir]]: ${EXTEN} hold the extension you are executing right at that line. |
| 20:21.47 | overyander | Chainsaw, the failed faxes fail after we negotiate the preamble and i send a CFR to the remote media server. I don't get anything back after a 10 second time-out |
| 20:21.54 | Cuzner | samdaman: it's truely the worst part of the job for a number of us in this channel. |
| 20:22.46 | [[thufir]] | exten => _9x.,1,Dial(${BABY}/${EXTEN:1}) should dial out, right? |
| 20:23.26 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 20:24.04 | Chainsaw | samdaman: I think you should *modem noises* |
| 20:24.13 | [TK]D-Fender | [[thufir]]: depends what YOU dial, and what's in that variable. Why waste time on a THEORETICAL question when you can just TRY it and SEE what it evaluates to? |
| 20:24.47 | [TK]D-Fender | [[thufir]]: Looks like you need to SEE the result to properly understand what you're even looking at. Because explaining the concepts alone doesn't seem to be sinking in that way. |
| 20:25.05 | igcewieling | [TK]D-Fender: I see nothing's changed. 8-| |
| 20:25.10 | Chainsaw | overyander: I would recommend that you try a physical device capable of T38. Asterisk itself will not terminate T38, merely allow it to traverse through. |
| 20:25.21 | [TK]D-Fender | igcewieling: Only the date on the calendar |
| 20:25.25 | Chainsaw | overyander: Trying to terminate within Asterisk itself without a physical phone line is a recipe for disaster. |
| 20:25.33 | Cuzner | [[thufir]]: NoOp() is your friend, if you're looking to see what the value of a varriable is. |
| 20:25.36 | igcewieling | Huh? |
| 20:25.50 | [[thufir]] | Cuzner: oh, didn't know about that. |
| 20:25.55 | igcewieling | That is the ONLY part of T.38 which works for us. PSTN -> T.38 -> Asterisk -> RxFax. |
| 20:26.12 | igcewieling | no other setup involving Asterisk has worked for us. |
| 20:26.58 | [[thufir]] | when exten => _9x.,2,Dial(${BABY}/${EXTEN:1}) is in context local_200 it's then passing that EXTEN variable to the context held by BABY? And then...? |
| 20:27.18 | [TK]D-Fender | no. |
| 20:27.23 | [TK]D-Fender | You dial DEVICES |
| 20:27.38 | [TK]D-Fender | do not call device definitions in sip.conf CONTEXTS |
| 20:27.47 | [TK]D-Fender | GO TRY IT NOW |
| 20:27.48 | Chainsaw | This is serious. The shift key is coming on. It'll be caps lock next. |
| 20:27.51 | samdaman | I think I have a T.38 passthrough setup .. I'm sending and receiving faxes through an ata but don't know if T.38 has anything to do with it, should it say something in the logs |
| 20:27.55 | igcewieling | Only takes me a few mins to remmeber why I left. Until next time [TK]D-Fender, be well. |
| 20:27.58 | Cuzner | [[thufir]]: ${EXTEN} will be whatever extension was matched on that particular line of dialplan. |
| 20:27.59 | *** part/#asterisk igcewieling (~ewieling@ip98-170-196-157.pn.at.cox.net) |
| 20:28.02 | [TK]D-Fender | Explanations are not sinking in. Go LOOK at what it does. This is wasting time. |
| 20:28.05 | Chainsaw | had better be quiet before Fender's restraining order kicks in |
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| 20:35.06 | [TK]D-Fender | packs up to head home |
| 20:35.09 | [[thufir]] | well, I gotta go. |
| 20:35.20 | [[thufir]] | lol, sunday I'll try again, time permitting. thanks. |
| 20:35.40 | eirirs | thufir hawat |
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| 20:49.10 | djgerm | Hello! I am trying to configure my asterisk (which I have setup very simply as a VOIP client for testing my main PBX) to use ICE, STUN, and TURN. I believe these configurations are made in rtp.conf, Do I just need to set icesupport=true, turnaddr=, and stunaddr=? |
| 20:50.39 | monsterco | Is there a tool I can use that would tell me number of simultaneous calls during a peak hour? I have a .csv sheet which has call duration, and time of call for each call inbound and outbound and would like to know how many channels I am using |
| 20:51.12 | robmal | Zabbix/nagios/cacti/munin |
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| 20:58.30 | monsterco | robmal - which one is easier to setup quickly? Also, do they eat .csv and spit out or do I have to do some sort of scripting? |
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| 21:01.19 | robmal | Munin will be the easiest, but very simple. It uses AMI, nagios uses asterisk CLI, the rest uses SNMP. |
| 21:01.53 | robmal | Each has a plugin for asterisk, so no problem there. |
| 21:02.26 | WIMPy | Except that you only ever get snapshots. |
| 21:03.14 | robmal | Yes. |
| 21:03.40 | WIMPy | That makes it pretty random. |
| 21:04.43 | robmal | The key factor was simplicity and easiness to install, not best quality. |
| 21:04.55 | robmal | Hell, with csv you can use excel to do it. |
| 21:05.51 | monsterco | robmal - this is just a .csv file now and not an active system |
| 21:05.55 | monsterco | it's history data |
| 21:06.20 | robmal | Oh, so go excel. |
| 21:07.15 | monsterco | robmal - excel you mean a chart? How can I direct it to check for time stamps? My main goal is to see what is the Maximum number of channels I ever used? basically intersection of time lines |
| 21:10.18 | robmal | Ok, so lets take the easy route. |
| 21:10.19 | robmal | http://www.voip-info.org/wiki/view/Asterisk+CDR+csv+mysql+import |
| 21:10.24 | robmal | http://www.cahilig.net/2008/06/10/how-install-web-based-asterisk-cdr-analyzer |
| 21:11.55 | trurl | the book i'm reading suggest "autoload=yes" and noload whenever needed. i don't know what modules are loaded and why. is this a valid suggestion or should i "autoload=no"? |
| 21:12.29 | Chainsaw | trurl: You can generally leave this to Asterisk. I would not attempt to micromanage the task. |
| 21:13.08 | trurl | Chainsaw: okay, then i'll follow the book |
| 21:13.24 | *** join/#asterisk CeBe (~CeBe@xd9bebe97.dyn.telefonica.de) |
| 21:13.27 | Chainsaw | trurl: I support that. autoload=yes and noload to veto any particularly poor choices. |
| 21:14.19 | trurl | i have moved every *.conf to ./samples and i'm copying back .conf files to get rid of all those errors when starting asterisk |
| 21:16.16 | monsterco | robmal - appreciate that but this is an edited file with only these fields and is not the original sheet from Asterisk (Date Stamp, Time stamp, Number, Duration) |
| 21:17.50 | robmal | Oh, you made it harder on purpose ;-) |
| 21:18.49 | monsterco | robmal - you got it :) |
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| 21:35.41 | *** join/#asterisk KungFuJesus (~adam@72.49.188.141) |
| 21:35.51 | KungFuJesus | Hello |
| 21:36.02 | KungFuJesus | We are using a TE134 card with the FreePBX distribution (http://freepbx.org). We are connecting to an Adtran appliance from Level3 which provides a soft PRI line. Whilst connected to this device, it works for a while, occasionally encountering latency bumps. Eventually, sometime between 24 and 72 hours, the card bumps in and out of the yellow alarm state. Eventually, we lose the ability to talk o |
| 21:36.08 | KungFuJesus | ver the PRI. We call Level3, they remotely turn the port off and back on again, and it starts working again. Level3 has told us itâs likely a problem on our end. They are willing to replace the Adtran appliance but if that doesnât resolve the issue then we are on the hook for the equipment. The TE134 is not sharing any interrupts and itâs the only PCI expansion card in the chassis. Doing |
| 21:36.14 | KungFuJesus | a cat /proc/dahdi/1 shows the span has had interrupt misses (not a huge number of them), CRC4 errors, and E-Bit errors. This is a single T1 span. We have not yet run the full loopback test in dahdi to determine if there are any hardware issues with the card. |
| 21:37.14 | KungFuJesus | We were told by Level3 that this can happen if the timing drifts out of sync enough. Sometimes when we'd try to reconnect after it went yellow it would say that the remote end thinks it is configured in CPE mode as well |
| 21:38.02 | Chainsaw | KungFuJesus: Since it is PCI (PCI-X, but still) do the old-school troubleshooting. |
| 21:38.17 | Chainsaw | KungFuJesus: Play PCI roulette until it works. Your logic board will have more slots. Try another one. |
| 21:38.43 | KungFuJesus | I've read that these newer cards are not nearly as sensitive to PCI latency anymore, though |
| 21:39.02 | Chainsaw | KungFuJesus: Additionally, look for the newest available BIOS. Do not get hung up on the change log, if you made some incredibly embarassing mistake you wouldn't list it either. |
| 21:39.06 | KungFuJesus | the current IRQ it's on doesn't appear to be shared with any other devices |
| 21:39.27 | Chainsaw | KungFuJesus: "appears" and "is not" are very different words. |
| 21:39.46 | KungFuJesus | well the procfs interface in /proc/interrupts says this |
| 21:39.58 | KungFuJesus | <PROTECTED> |
| 21:40.13 | WIMPy | That just means that there's no other driver listening on that IRQ. |
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| 21:40.24 | Chainsaw | KungFuJesus: 1) PCI roulette 2) BIOS upgrade 3) TE134 RMA 4) AdTrans replacement 5) Asking #asterisk again |
| 21:40.30 | WIMPy | It doe NOT mean there's no other device geerating them. |
| 21:41.06 | Chainsaw | Let us know when you hit 5. Hint: You probably won't. |
| 21:41.35 | WIMPy | And timing erros must not accumulate. If they do there's something wrong somewhere. |
| 21:42.03 | KungFuJesus | unfortunately this troubleshooting will necessitate a multi-day affair |
| 21:42.06 | Chainsaw | Timing matters in ISDN. You snooze, you lose. |
| 21:42.19 | Chainsaw | KungFuJesus: Well you can do it fast or you can do it properly. |
| 21:42.20 | Chainsaw | KungFuJesus: Which would you prefer? |
| 21:42.32 | KungFuJesus | You'd think modern PCI controllers and hardware wouldn't have high latencies |
| 21:43.08 | KungFuJesus | Are the PCI-Express versions of these cards immune to these issues? |
| 21:43.12 | Chainsaw | KungFuJesus: Where I come from we call that naivety. |
| 21:43.20 | Hsilamot | anyone here can help me to determine if this is a bug? |
| 21:43.24 | Chainsaw | KungFuJesus: No, they just hide them behind a PCI-PCIe bridge. |
| 21:43.40 | KungFuJesus | But it's not a shared bus |
| 21:43.49 | Hsilamot | i've got an asterisk installation with freepbx and i have both LAN and WAN clients |
| 21:44.12 | Hsilamot | because of the WAN attempts to reach my pbx i changed the default port for asterisk |
| 21:44.17 | Chainsaw | Hsilamot: If what is a bug? I never saw an original enquiry. |
| 21:44.43 | Hsilamot | but now the WAN clients can't control the calls, they register, start the call and will break the SIP connection |
| 21:44.45 | trurl | okay, this bleeding edge 13.3 doesnt match my book and i realize that i _do_ like debian after all. is 11.13 recent enough? or am i going to miss out something? |
| 21:45.13 | Hsilamot | asterisk is sending this header: Contact: <sip:*999@187.178.XXX.XX:5060;transport=TCP> |
| 21:45.14 | Chainsaw | KungFuJesus: Yes it is. |
| 21:45.23 | WIMPy | trurl: you miss pjsip. |
| 21:45.29 | Chainsaw | KungFuJesus: That "internal NIC" of yours is PCI. |
| 21:45.30 | Hsilamot | with the 5060 port in it, which is wrong |
| 21:45.52 | KungFuJesus | Not using it, it's disabled in the BIOS and a PCI-Express NIC is being used instead |
| 21:46.06 | Chainsaw | KungFuJesus: Which may well have a PCI to PCIe bridge. |
| 21:46.09 | Hsilamot | but in the internal calls it does send the correct port Contact: <sip:*999@10.1.7.1:6506;transport=TCP> |
| 21:46.18 | Chainsaw | KungFuJesus: Additionally the BIOS disabling the NIC may not stop linux enumerating it. |
| 21:46.41 | trurl | WIMPy: mh. i dont know what that is ;) (yet) |
| 21:46.49 | KungFuJesus | it is invisible according to lspci |
| 21:47.37 | Chainsaw | KungFuJesus: Still doesn't mean the bus isn't shared. There will be PCI to PCI bridges involved. |
| 21:47.47 | Chainsaw | KungFuJesus: On the 5 step plan, you remain on step 0. |
| 21:50.06 | KungFuJesus | I'll try it, and I hope you're right. This doesn't seem like an issue that should be happening, though. |
| 21:51.01 | WIMPy | Correct. It shouldn't happen, but PC hardware has always been extremely crappy. |
| 21:51.15 | Chainsaw | KungFuJesus: There is firmware involved, which is written by humans. Humans make mistakes, then cover them up. Assuming perfection is ill-advised. |
| 21:51.36 | KungFuJesus | There's also Digium authored firmware involved |
| 21:51.53 | KungFuJesus | is there any way to get more diagnostic messages from the card? What's spat out is a bit limited |
| 21:51.58 | Chainsaw | My lawyer advises that I refrain for commenting about whether Digium & perfection ought to go together. |
| 21:52.27 | WIMPy | Watch the number of errors the driver reports. |
| 21:52.45 | KungFuJesus | at one point we say this guy: chan_dahdi.c: PRI Span: 1 Write to 42 failed: Unknown error 500 |
| 21:52.49 | KungFuJesus | saw* |
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| 21:53.32 | KungFuJesus | WIMPy: and in terms of errors, do you mean what the dahdi proc interface is reporting? |
| 21:53.44 | WIMPy | Or you try to hold of some professional testing equipment to analyze timing and jitter. |
| 21:53.49 | WIMPy | yes |
| 21:55.04 | KungFuJesus | just curious, how effective are the loopback tests for dahdi at finding these sorts of issues? If I do have an issue with PCI latency, will the tests fail? |
| 21:56.10 | Chainsaw | They may, eventually. |
| 21:56.15 | WIMPy | If you use a loopback plug on the card you would see issues up to the CPU, so that definitely includes interrupt handling. |
| 21:56.20 | KungFuJesus | ah, just a matter of how long I run the loopback test |
| 21:56.36 | Chainsaw | Certainly wouldn't attempt to draw conclusions from a 10 second run. |
| 21:57.07 | KungFuJesus | Of course not. Unfortunately performing this tests will require that I do it during non-peak hours |
| 21:57.12 | KungFuJesus | test* |
| 21:57.42 | Chainsaw | KungFuJesus: You can use dmidecode and/or biosdecode to check your current BIOS version and see whether a new one is available. |
| 21:58.01 | Chainsaw | KungFuJesus: I would queue up what you can for the maintenance windows, for they are few and closely guarded. |
| 21:58.13 | WIMPy | IT's always good to have more ports than you need. |
| 21:58.48 | KungFuJesus | this is the board: http://www.asus.com/Motherboards/M4A77D/ |
| 21:59.48 | Chainsaw | KungFuJesus: Sounds like a gaming/desktop board. |
| 22:00.00 | KungFuJesus | indeed it is, unfortunately it was what was in the office |
| 22:01.02 | Chainsaw | KungFuJesus: So your Ferrari doesn't get good town mileage. You may be using it outside of its target market. |
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| 22:46.50 | averythomas | I have a funny story about a call center using * |
| 22:47.33 | averythomas | One of my friends got a call from a telemarketer but my friend told him to transfer him to "type this code in" |
| 22:47.43 | averythomas | the guy typed the code in that reset the whole phone system :P |
| 22:58.42 | newtonr | wut |
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| 23:19.43 | WIMPy | Who on earth would have a code to reset the whole system to be used by anyone? |
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| 23:24.04 | newtonr | WIMPy, and why would a telemarketer transfer the customer to X extension without knowing what that extension is? |
| 23:24.46 | WIMPy | Well, if he was that clever, he would probably have another job. |
| 23:35.22 | newtonr | touche |
| 23:36.05 | djgerm | Either that or be in jail. (it's not the 90s anymore, breaching security like that just makes companies mad nowadays) |
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| 23:41.09 | [TK]D-Fender | is it a breach of security if you simply dial an extension that your device clearly has the rights to? |
| 23:41.39 | WIMPy | Sure. But not by the user doing it. |
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