| 00:13.12 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
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| 00:35.15 | *** join/#asterisk Free99 (~Free99@cpe-66-108-105-10.nyc.res.rr.com) |
| 00:35.56 | Free99 | hey guys. I've been trying to figure out how to do this for the past few hours, and what I've come up with has got to be more easily done in a simple way |
| 00:36.42 | Free99 | If you're listening for someone to dial a number for outside access (I can't use DNID), how would you strip a 1 off the entered number if it is there? |
| 00:37.18 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 00:37.49 | WIMPy | ${EXTEN:1} |
| 00:37.59 | WIMPy | Or what exactely are you trying to do? |
| 00:38.03 | Free99 | I basically have Read(inputvar,"silence/1",,,,10), and then I'm trying to use execif |
| 00:38.14 | Free99 | strip a 1 off the entered number if its there |
| 00:38.34 | Free99 | a 1 prefix, to be precise |
| 00:39.02 | WIMPy | Well, then execif is the obvious one. |
| 00:39.57 | Free99 | WIMPy, yep. thing is, do I really need to jump to a second label or something if I just want to execute a simple action like Set(varname=${varname:1})? |
| 00:40.28 | Free99 | would it help if I pastebin the relevant section? |
| 00:40.32 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 00:40.37 | WIMPy | ANo, that would be GotoIf. With ExecIf you just do it there. |
| 00:41.43 | Free99 | so same => n,ExecIf($["${GET:0:1}" = "1"]?Set(varname = ${varname}) looks ok? |
| 00:42.06 | Free99 | rather, same => n,ExecIf($["${GET:0:1}" = "1"]?Set(varname = ${varname:1}) |
| 00:42.12 | WIMPy | If you add a :1 near the end. |
| 00:42.21 | WIMPy | Yes, that way. |
| 00:42.32 | Free99 | dude awesome. thanks very much |
| 00:43.53 | Free99 | WIMPy, by the way: how do I know when I need to quote things and such, or is it similar to sh programming, where "" forces something to be parsed as a string |
| 00:44.27 | WIMPy | No, Asterisk doesn't use quotes. They become part of both strings. |
| 00:45.31 | WIMPy | But usingg either quotes or anything else prevents syntax errors in case a variable is empty similar to what happens on the shell. |
| 00:45.47 | Free99 | hmm. so why is it necessary to, in a execif for instance, use "${varname}" when comparing two items? Based on what I've read it looks like it has something to do with handling null input |
| 00:46.36 | WIMPy | Exactely. You can do it the same way as some people do in the shel as well and compare X{varname}=X0. |
| 00:46.49 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 00:47.02 | WIMPy | Quotes just look nicer, but don't have a special meaning. |
| 00:47.29 | Free99 | Good to know :) thanks WIMPy |
| 00:50.37 | igcewieling | Free99: your question indicates a basic lack of understanding about Asterisk, You should read the book. what you are doing is covered in the book. |
| 00:50.39 | igcewieling | ~book |
| 00:50.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 00:52.01 | Free99 | whoa, igcewieling you stopped ignoring me. Thanks for the tip |
| 00:52.22 | igcewieling | Free99: My /ignores are not persistant. |
| 00:52.46 | igcewieling | ok, MOST of them are not. |
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| 00:59.39 | Free99 | asterisk is so damn cool lol |
| 01:00.14 | leifmadsen | it's ok |
| 01:00.15 | leifmadsen | :D |
| 01:00.52 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 01:01.38 | Free99 | is it true that one is required to retain voicemail for up to 7 years if you're an international service provider? The president of the company keeps mentioning that but I find no record of it |
| 01:02.05 | WIMPy | Sometimes it feels like I'm one of a few who have missed the transition to FreeSwitch. |
| 01:02.06 | Free99 | so we've decided not to offer voicemail though I think people would really like it |
| 01:04.08 | leifmadsen | pffft freeswithc |
| 01:04.13 | leifmadsen | I guess if you love xml |
| 01:04.33 | WIMPy | No, I hate it. |
| 01:05.05 | Free99 | yeah, but trying to maintain an asterisk config via database is a bitch if you don't want to expose yourself with phpmyadmin or w/e |
| 01:05.17 | Free99 | they all suck respectively |
| 01:05.24 | Free99 | :P |
| 01:06.04 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 01:10.16 | leifmadsen | well you shouldn't be maintaining any database by hand :) |
| 01:10.22 | leifmadsen | you're doing it wrong./ |
| 01:12.11 | Free99 | good thing I'm not... but if you've run a DB long enough, you know at some point you'll need to peer into it or modify a value manually. Not going to say its a guarantee, but.. it might be lifetime warrantied |
| 01:12.27 | Free99 | (shrug) iunno man, that's all I got |
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| 01:23.35 | gnudna | hi guys |
| 01:24.57 | gnudna | anybody using asterisk from debian backports and has cisco 7960 phones? |
| 01:25.15 | gnudna | they are failing registration but i am able to dial out |
| 01:25.26 | gnudna | calling the phone takes me to voicemail |
| 01:25.50 | gnudna | anybody have a simple solution for this while staying on the version of asterisk i am on? |
| 01:26.42 | gnudna | version i am on btw is 1.8.13.1~dfsg-1~bpo60+1 |
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| 01:56.21 | gnudna | so nobody has any thoughts? on getting a cisco 7960 to register with asterisk 1.8? |
| 01:57.26 | gnudna | sip debug shows the phone sending register request but asterisk does not seem to be honoring it? |
| 01:58.44 | gnudna | pastebin link or sip debug http://pastebin.com/JsAbDHYh |
| 01:58.57 | gnudna | btw this worked on asterisk 1.6.x |
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| 02:00.26 | gnudna | all cisco 7960 are doing this since upgrade ;( |
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| 02:42.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 05:38.15 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 05:38.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 05:53.21 | [TK]D-Fender | Addisk: Everyone who I've ever sen come in here with those had loads of problems |
| 05:57.12 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 05:58.36 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 06:05.48 | bulkorok | hi |
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| 06:31.05 | *** join/#asterisk toresbe (~toresbe@158.36.190.254) |
| 06:31.45 | toresbe | Hey folks. I have a non-strictly-Asterisk question about an odd piece of hardware I've found three of: It's a Codex 6250 E1 multiplexor. |
| 06:32.17 | toresbe | Does that ring any bells with anyone here? |
| 06:32.40 | ChannelZ | Only the ringing in my ears |
| 06:32.56 | toresbe | I've googled but found very little, certainly not enough for someone as inexperienced as I am with telephony to piece together the necessary info |
| 06:34.06 | ChannelZ | What is the actual question? |
| 06:34.46 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.240) |
| 06:34.59 | toresbe | I'm trying to phrase it well - basically, the device has some E&M and some FXS ports, and muxes them over an E1. I guess my question would be - how can I tell if it will play well with Asterisk? |
| 06:36.32 | ChannelZ | Well I'm guessing you'd need an E1 card on the Asterisk side but after that no reason why not |
| 06:37.10 | ChannelZ | IE this thing is a black box with an E1 port on it? (plus all the FXSes and such) |
| 06:40.47 | Addisk | FML, did i get a bust with redfone device? |
| 06:42.06 | ChannelZ | So far yes |
| 06:44.12 | ChannelZ | It seems troubling that their setup support refers to Asterisk 1.4 and 1.6 |
| 06:45.25 | ChannelZ | But it looks like TDMoE which is more agnostic (a DAHDI-side thing) |
| 06:46.23 | toresbe | ChannelZ: yep. I'd get the hardware, of course. My worry is just about the possibility that this might have some different sort of encapsulation or multiplexing standard that would prevent Asterisk from seeing the lines. |
| 06:48.18 | ChannelZ | well technically speaking E1/T1 et all are defined standards, so if these multiplex boxes speak their own language I'm not sure what they're useful for. |
| 06:49.48 | toresbe | point-to-point links, maybe? This comes from a TV station, so there's tons of unconventional equipment in use - as well as conventional equipment in unconventional setups :) |
| 06:50.42 | ChannelZ | TV is nothing if not unconventional |
| 06:51.03 | ChannelZ | But E1 should be E1 and if it's not they shouldn't be calling it that. |
| 06:52.30 | ChannelZ | You might have to sit around here for weeks asking in order to find someone who has that particular kit. You're better off trying to find someone local to you with a Digium card you can test with and just try it, or buy one and if it doesn't work resell it for a few bucks loss |
| 06:52.56 | Addisk | damn |
| 06:53.26 | Addisk | well its brand new, and been < a month |
| 06:53.27 | ChannelZ | (that was to toresbe BTW) |
| 06:54.04 | ChannelZ | Addisk: eh? |
| 06:59.33 | Addisk | the redfone devices i got are new |
| 06:59.47 | Addisk | i got the 2 port & 1 port, hoping it can utilize them in a virtual asterisk setup |
| 07:00.35 | Addisk | cent 6.4 is hyperv ready, so i got that going on a hyperv machine working. just can't configure the damn redfone |
| 07:00.40 | Addisk | same issue with virtualbox |
| 07:01.01 | Addisk | if this is no good, then i gotta fall back to my old phsyical + t1 card machines |
| 07:01.14 | ChannelZ | well since they are TDM over Ethernet they do need proper networking access |
| 07:01.23 | Addisk | the access works |
| 07:01.35 | Addisk | just the tool doens't seem to match their documentation |
| 07:01.42 | Addisk | like setting the IP & configuring the dadhi |
| 07:01.58 | Addisk | when i try to match the dahdi setting, i get a system halt error, and my vm is gone |
| 07:02.05 | Addisk | i have to revert to previous setting so i can get back into it |
| 07:02.44 | Addisk | i'm gonan try to contact their support, but hopeing someone here would know |
| 07:04.47 | ChannelZ | I believe the DAHDI side is by MAC address not IP |
| 07:06.13 | Addisk | yea |
| 07:06.20 | Addisk | i'm doing the config via mach |
| 07:06.22 | Addisk | mac* |
| 07:06.50 | Addisk | so i added dynamic=ethmf,eth1/00:50:C2:65:XX:XX/0,31,0 into the dahdi/system.conf |
| 07:07.07 | Addisk | and the bchan=1-23, dbdan=24 |
| 07:07.13 | Addisk | and the system dies |
| 07:07.19 | Addisk | when dahdi reloads |
| 07:08.50 | ChannelZ | I can only guess that it's the VM barfing on whatever lower level network access the TDMoE driver is trying to do. Or the modules are bunk. |
| 07:09.27 | ChannelZ | Not sure how VM-tollerant DAHDI is. Did you compile it yourself? |
| 07:09.33 | Addisk | nope |
| 07:09.49 | ChannelZ | so it's a package? |
| 07:09.50 | Addisk | so i tried out piaf, they actualy setup the entire image for testing |
| 07:09.51 | Addisk | yea |
| 07:09.55 | Addisk | for virtualbox |
| 07:10.51 | ChannelZ | shrugs |
| 07:11.08 | Addisk | i can tell you my version is 2.6.2 |
| 07:11.17 | Addisk | while the redfone refers to 2.4 |
| 07:11.56 | ChannelZ | Out of my realm. It sounds like DAHDI that is crashing, not related to the redfone in particular. VM issue |
| 07:16.51 | Addisk | whats the usual command to reload the dadhi setup? |
| 07:17.16 | Addisk | i've been doing dadhi_cfg, genconf, scan |
| 07:19.04 | Addisk | omg, one of those worked lol |
| 07:19.19 | *** join/#asterisk angryuser_laptop (~angryuser@ip-189.net-81-220-67.lyon.rev.numericable.fr) |
| 07:19.24 | Addisk | system didn't halt, and dahdi reloaded |
| 07:20.16 | Addisk | does the dahdi_test bring much revlanence anymore? |
| 07:20.37 | Addisk | i remember back when i setup my super old asterisk 1.2, it was a biggy |
| 07:20.47 | Addisk | the system is getting 99.99x |
| 07:21.10 | Addisk | with this vm, i'm getting like 99.8xx |
| 07:21.11 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:23.40 | Addisk | yup there we go, a manual config with 2 lines bust the VM, fml |
| 07:23.46 | Addisk | 2 T1 |
| 07:25.14 | Addisk | what does it mean, "TDMoX: No master." |
| 07:27.16 | Addisk | damnit it all, should of gotten the SIP gateways |
| 07:30.36 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 07:33.31 | *** join/#asterisk Rhomber (~david@60-240-245-17.static.tpgi.com.au) |
| 07:33.33 | kaldemar | Addisk: DAHDI does not have a master clock. from system.conf.sample: "Note that you MUST have a REAL DAHDI device if you are not using external timing." |
| 07:33.45 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:33.54 | kaldemar | Addisk: you have timing set to 0 which means you're not using external timing. at least for the dynamic span. |
| 07:35.06 | Addisk | what should i have there? |
| 07:35.30 | Addisk | well, when it says real, does it mean i need a real PCI/PCIe dahdi card? |
| 07:35.44 | Addisk | i was being told that the redfone emulates itself to be in teh systme somehow |
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| 07:42.11 | Addisk | um, i set the other one to be 1 |
| 07:42.24 | Addisk | and get half after dahdi_cfg |
| 07:45.17 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 07:48.32 | Addisk | damnit it all i guess i need some sleep and figure this out in morning :( |
| 07:50.07 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 07:50.14 | eagles0513875 | Addisk: usually when you leave stuff alone they have a strange way of working themselves out when you come back to them |
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| 08:00.38 | fling | kaldemar: why do I need to reboot my nat router to fix issues with sip? |
| 08:01.01 | fling | kaldemar: registration starts working and stun peer become online after reboot |
| 08:01.22 | fling | kaldemar: no reboots needed for connecting for other peers :| am I missing something? |
| 08:01.33 | fling | like route cache or something else… |
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| 08:07.31 | kaldemar | fling: your router is crap. |
| 08:07.58 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:09.19 | ChannelZ | FOR ME TO POOP ON! |
| 08:14.07 | fling | kaldemar: ok :P |
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| 08:22.48 | polysics | does the "quiet" option on a peer mean he will not HEAR ConfBbridge announcements, or PRODUCE any? |
| 08:23.03 | polysics | I would like admins to hear "someone joined" sound while normal users should not |
| 08:23.21 | polysics | but it looks like quiet is "do no produce sounds when joining", so the reverse |
| 08:31.14 | polysics | I hate convoluted customer requests. |
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| 08:52.08 | polysics | no one knows? :-) |
| 08:53.34 | Faustov | sorry I'm still stuck on meetme ;) |
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| 10:03.06 | linocisco | hi all |
| 10:03.31 | emamdouh | <PROTECTED> |
| 10:04.12 | linocisco | how can users generally know which PSTN lines are busy or free before making a phone call outside? |
| 10:05.02 | kaldemar | linocisco: "PSTN line" really means nothing. and a user probably has no such information. your PBX might have. |
| 10:06.07 | linocisco | kaldemar, so to save time for users, how can users know which lines are free or busy before entering all long no. ? |
| 10:06.17 | kaldemar | emamdouh: see what lsdahdi tells you |
| 10:07.55 | kaldemar | linocisco: maybe you'd need to build a web page that shows status or something similar. sounds quite dumb though. if they have time to stare at such a status, they probably have time to dial the number aswell. |
| 10:08.58 | linocisco | kaldemar, no. what I mean is that there can be two confusing "busy status", all outgoing PSTN lines are busy or destination no. are busy |
| 10:09.24 | linocisco | kaldemar, if either one is busy , call cant be established directly |
| 10:10.16 | linocisco | kaldemar, users should have line indicator or something before making calls to see "busy" status is from which side |
| 10:11.42 | kaldemar | linocisco: you only know the status of your connectivity, if the interface provides such status. you don't have any information about the destination. |
| 10:12.34 | linocisco | kaldemar, how can I know status of connectivity as normal user via phone? |
| 10:12.54 | kaldemar | what kind of a phone? |
| 10:18.32 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 10:19.44 | linocisco | kaldemar, cisco 7942G and SPA502G and ATA FXS HT702 with 2 ports |
| 10:23.03 | kaldemar | deja vu. use device state or keep count in your dialplan. |
| 10:23.36 | *** join/#asterisk stoptheruckus (~youjelly@39.47.204.36) |
| 10:23.57 | kaldemar | anyway, you basically cannot indicate the state in your phone without making a call. |
| 10:25.02 | kaldemar | some phones support so called desktop messages that stuff text in the idle screen of the phone. you'd need to build something to send those if your phone(s) support them. |
| 10:26.48 | linocisco | kaldemar, in real world asterisk deployment, how do u all setup for this kind of line information? |
| 10:31.27 | kaldemar | i'd guess there aren't many who do. |
| 10:32.22 | kaldemar | if your users get busy because you don't have enough capacity, there are better solutions. like getting more capacity. |
| 10:36.36 | linocisco | kaldemar, no. i mean just to want to give users know the status before making calls |
| 10:36.57 | linocisco | that feature can be easily avaliable on Panasonic Legacy PBX |
| 10:39.18 | *** join/#asterisk MrNemo (~MrNemo@80.84.58.91) |
| 10:44.06 | MrNemo | ребят, подскажите хороший гайд по вытаскиванию cdr, желательно со скриптами оптимизации удобочитаемости, если таковой имеется, не смог найти что-то более менее путное |
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| 10:58.50 | emamdouh | kaldemar: check http://pastebin.com/r1TUwP8W |
| 10:59.00 | emamdouh | kaldemar: what does this mean ? |
| 10:59.52 | kaldemar | emamdouh: you have red alarms on channels 1-3 which in your case probably means that they are not connected. |
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| 11:00.14 | *** join/#asterisk rox (~rox@BSN-142-168-6.dial-up.dsl.siol.net) |
| 11:00.45 | rox | greetings |
| 11:00.47 | emamdouh | kaldemar: ok thanks |
| 11:02.31 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 11:03.17 | rox | I have a SIP peer, that has qualify=no (it doesn't take OPTIONS packets), however, i still need to fail over to another peer if this peer is down. Is there a way of doing that in Dial()? From what i read in the manual, Dial() can timeout if the call isn't answered, but does anybody know of an option for Dial() to timeout if ringing doesn't start in a specified period? |
| 11:04.09 | WIMPy | Look at the timers in sip.conf. |
| 11:05.55 | rox | WIMPy: what specifically did you have in mind? |
| 11:06.58 | WIMPy | Nothing |
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| 11:15.07 | rox | Regarding SIP session timers; If I set the session-expires parameter to something like 2 seconds, then Dial() will fail after that period of time? |
| 11:16.29 | WIMPy | You can't set session-timers that low. |
| 11:16.54 | WIMPy | The timers should be called timer* AFAIR. |
| 11:18.04 | polysics | does anyone know what "quiet" in ConfBridge actually does? thanks :-) |
| 11:18.18 | polysics | does it prevent the user from HEARING conf sounds or from MAKING them? |
| 11:20.24 | rox | WIMPy: session timers will not solve my problem. Thank you for your suggestion, though, I appreciate it. |
| 11:21.39 | WIMPy | It's the wrong timer. |
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| 11:24.22 | kaldemar | rox: not session timers, "SIP timers" |
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| 11:33.43 | polysics | finding docs on some functions is quite difficult :-) |
| 11:33.54 | rox | kaldemar: i checked it out. I guess i need to use timerb=timeout. Do you perhaps know, will this exit Dial() after timeout even if qualify=no is set? |
| 11:34.06 | kaldemar | polysics: "core show function <function>" |
| 11:34.38 | kaldemar | rox: qualify has little to do with Dial behavior. |
| 11:35.21 | rox | kaldemar: well, if qualify=no is set, Dial() will hang forever if the peer does not respond. |
| 11:35.25 | polysics | kaldemar: the "quiet" confbridge.conf value (and some others) seem to not be actually explained anywhere |
| 11:35.40 | polysics | unless I am dumb but I still don't get the explanation :-) |
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| 11:36.23 | kaldemar | polysics: it has some kind of explanation in the sample config. |
| 11:37.13 | polysics | kaldemar: do you happen to know, by chance, if quiet means "you hear no sounds" or "you don't make sounds when entering etc"? |
| 11:37.16 | Greenlight | Doesn't quiet just stop the leave and join sounds? |
| 11:37.19 | polysics | that's what is not clear |
| 11:37.29 | Greenlight | It's what you *hear* |
| 11:38.09 | polysics | so, if A is quiet=no and is in the conference, and B is quiet=yes and joins, A hears the sound (provided you configure it ofc) and B does not? |
| 11:38.10 | Greenlight | Afaik there's no way to have certain people not generate the sounds and others to generate them, unless of course you implemented something yourselfg |
| 11:38.34 | Greenlight | polysics: Exactly |
| 11:38.45 | Greenlight | It controls what the profile will hear |
| 11:38.53 | polysics | then I am doing something wrong, but thanks for clarifying that :-) |
| 11:38.59 | Greenlight | No probs |
| 11:39.08 | WIMPy | Doing it yourself isn't a bad idea as you can do it in parallel then. |
| 11:39.23 | polysics | I am probably not setting the correct profile for either of the users |
| 11:39.47 | polysics | WIMPy: this app already has a ton of "outside" logic, everything I can get from "static" config I would like to have |
| 11:40.10 | WIMPy | You can also set specific options via the dialplan function. |
| 11:40.41 | polysics | I tried both but I am not hearing the sound - investigating what's wrong :-) |
| 11:40.49 | kaldemar | polysics: quiet=yes means "do not play join/leave sounds to user or conference" |
| 11:41.20 | polysics | kaldemar: so if an user with quiet=yes joins, he does not "make" sounds for the others either? |
| 11:41.26 | polysics | sort of a ninja mode? |
| 11:41.36 | polysics | that's the reverse of what Greenlight above said |
| 11:41.47 | Greenlight | Yea, that's not how I've seen it work |
| 11:41.52 | Greenlight | Or how I understood it |
| 11:42.01 | Greenlight | quiet=no and they hear leave and join sounds |
| 11:42.30 | kaldemar | polysics: quiet=yes should not play join sound to the conference, yes. |
| 11:42.48 | polysics | hmm, ok |
| 11:42.51 | kaldemar | polysics: i didn't test it, just took a look at the code briefly. |
| 11:42.56 | polysics | what would get me the above behavior then? |
| 11:43.06 | polysics | A is in conf, B joins, A hears "ping" sound but not B |
| 11:43.11 | polysics | if there is something ,that is |
| 11:43.26 | kaldemar | polysics: modifying the source. :P |
| 11:43.27 | Greenlight | That's not the way the docs read either: When set to "yes," enter/leave prompts and user introductions are not played. By default, no. |
| 11:43.41 | polysics | are not played is ambiguous |
| 11:44.02 | polysics | kaldemar: no ConfBridge config does the above? Knowing is half the battle, thanks :-) |
| 11:44.05 | kaldemar | polysics: both "plays" are inside if (!quiet). and quiet is the flag in the user profile. |
| 11:44.14 | polysics | Adhearsion to the rescue |
| 11:44.46 | Greenlight | I guess it's simple enough to test |
| 11:45.47 | polysics | yeah, I am not hearing the sounds |
| 11:45.58 | polysics | BUT I was unsure I was doing things correctly |
| 11:46.00 | polysics | now I know I am |
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| 11:47.13 | polysics | ok, another difficult one |
| 11:47.28 | polysics | can I set Caller ID so that A sees B's number on the phone LCD? |
| 11:47.40 | polysics | caller ID or whatever allows me to set that message |
| 11:48.02 | WIMPy | That question needs clarification. |
| 11:48.33 | WIMPy | Do you want to do it in the conference? |
| 11:48.55 | polysics | yes, in the conference |
| 11:48.59 | Greenlight | polysics: For a normal call that's what you'd normally see. If you mean for the conference, then it depends on your situation |
| 11:49.12 | polysics | A is alone in the conf. B joins. A sees B's number on the LCD/softphone. |
| 11:49.19 | WIMPy | You have to do it manually using CONNECTEDLINE. |
| 11:49.23 | Greenlight | What if A isn't alone |
| 11:49.38 | polysics | A will always be alone or with B, confs are limited to 2 |
| 11:49.47 | polysics | we use this to lower connection times for an outbound call center |
| 11:49.54 | Greenlight | If Confs are limited to 2, why not use a normal call |
| 11:49.59 | WIMPy | Maybe you should use Bridge instead of ConfBridge? |
| 11:50.04 | Greenlight | +1 |
| 11:50.06 | polysics | lower connection times |
| 11:50.17 | polysics | oh, Bridge, I see |
| 11:50.25 | polysics | I could park people on MoH |
| 11:50.30 | polysics | agents, I mean |
| 11:50.35 | polysics | and just bridge them |
| 11:50.46 | polysics | yeah, that's an idea to consider |
| 11:50.47 | Greenlight | If this is for outbound dialling, stay clear of what your trying to do with ConfBridge, I speak from experiance, it's a world of pain :) |
| 11:51.24 | polysics | we are 90% done :-) |
| 11:51.31 | polysics | you would use MoH + Brdige? |
| 11:51.43 | polysics | how do you return agent to MoH after call is done? |
| 11:51.44 | Greenlight | I've moved from the ConfBridge method to that, yes |
| 11:52.00 | Greenlight | Bridge does that, it dumps Channel1 back into dialplan |
| 11:52.18 | Greenlight | You just need a Wait(600) then a GoTo to loop it |
| 11:52.29 | Greenlight | After the bridge, the agent goes back into their "holding area" |
| 11:52.52 | polysics | that sounds way more sensible |
| 11:53.02 | Greenlight | The main problem you'll get with confbridge is that you're very reliant on really good timing on asterisk |
| 11:53.09 | Greenlight | It's *mixing* EVERYTHING |
| 11:53.11 | polysics | well, let's pretend I can't change for now. CONNECTEDLINE sets caller ID arbitrarily? |
| 11:54.37 | WIMPy | It obviousely depends on the channeltype used, but generelly, yes. |
| 11:55.01 | polysics | SIP only here |
| 11:55.23 | polysics | so it is as simple as EXEC CONNECTEDLINE "Poly<1234>"? |
| 11:56.03 | WIMPy | The syntax doesn't look good. |
| 11:56.10 | WIMPy | It's a function. |
| 11:56.38 | Greenlight | Is this via AGI ? |
| 11:56.47 | polysics | either AGI or AMI |
| 11:56.51 | polysics | AMI is what I prefer |
| 11:57.02 | Greenlight | Set it, like a variable |
| 11:57.06 | Greenlight | Since it's functiomn |
| 11:59.16 | polysics | so it would use AMI SetVar? I don't think I get it, sorry |
| 11:59.53 | WIMPy | yes |
| 12:00.28 | Greenlight | Exactly, let me grab an example |
| 12:00.37 | Greenlight | AMIRequest_SetVar(oCurrentChannel, "AUDIOHOOK_INHERIT(MixMonitor)", "yes"); |
| 12:01.08 | Greenlight | In this example AUDIOHOOK_INHERIT is the function, and we SetVar it to "yes" |
| 12:01.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:01.49 | WIMPy | And you need that function we don;t know :-) |
| 12:02.04 | polysics | in waht sense we don't know? :-) |
| 12:02.52 | WIMPy | We don't know that function. We can guess it, but I question it's an example. :-) |
| 12:03.36 | Greenlight | Yea, I was relying on your guessing that all it does is do a SetVar :) |
| 12:03.57 | Greenlight | Channel, Variable, Value ... |
| 12:04.36 | polysics | Variable would be, say, CONNECTEDLINE(number,i) |
| 12:04.41 | polysics | and value 123456 |
| 12:04.51 | polysics | does that work? |
| 12:05.25 | Greenlight | Sure, that looks good |
| 12:05.43 | WIMPy | If another one without ,i follows, yes. |
| 12:06.04 | Greenlight | I just stops spamming messages doesn't it |
| 12:06.06 | Greenlight | "i" |
| 12:06.25 | Greenlight | Or does that actually stop the message to the handset ? |
| 12:06.30 | polysics | apparently, the docs are sparse on this too |
| 12:06.58 | Greenlight | Yea, after looking at the docs there, loose the "i" |
| 12:07.04 | WIMPy | The idea is if you want to set num and name, you set the first with ",i", preventing it from doing anyhting and then the oter without ",i" which then sends out both. |
| 12:08.26 | WIMPy | You could do both without ",i", but some phones don't like to be spammed. |
| 12:09.44 | WIMPy | I think it would have been clearer if you would just set variables and then call an application to send the update, but off course the current way saves the extra step. |
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| 12:17.50 | tparcina | Can I use one manager user for more clients? |
| 12:19.10 | tparcina | I'm planing to setup Nojee Click 2 dial, and I wonder can more users use same manager user? |
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| 12:19.24 | [TK]D-Fender | tpiif you mean AMI, yes |
| 12:19.31 | [TK]D-Fender | tparcina: if you mean AMI, yes |
| 12:19.41 | WIMPy | Even if I don't know what you want to do, I say yes, as I'm sure it's the answer to any possible meaning of the question. |
| 12:19.59 | tparcina | [TK]D-Fender: Thank you. |
| 12:20.36 | tparcina | WIMPy: Thank you as well. :D |
| 12:21.34 | polysics | great, thanks |
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| 12:23.41 | tparcina | To dial two numbers using manager interface, call, command and user rights should be enough, right? |
| 12:25.55 | tparcina | I see plenty of configuration examples giving, simple click2dial Firefox extension that uses manager interface, all rights. But I assume those (call,command,user) should be enough. |
| 12:32.50 | [TK]D-Fender | tparcina: No, "command" is deadly. Do not give it to anything you don't have to. |
| 12:33.22 | Kobaz | you dont need user either |
| 12:33.34 | cerienjean | Hello - I am looking for advice on multiwan/multihomed installation. I'd like to have one asterisk server, with multiple nics, each nated (no choice), with a dsl link and a public adress. How can I have clients registering on either nic and asterisk replying thorugh the correct nic ? |
| 12:33.58 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-niqabjcfvahebqup) |
| 12:34.25 | tparcina | Call will be enough? |
| 12:34.26 | [TK]D-Fender | cerienjean: You'd have to run separate SIP proxies for each nic. |
| 12:34.33 | [TK]D-Fender | tparcina: To place a call? yes. |
| 12:34.52 | cerienjean | that is what I was starting to think - what proxy could you recommend - I guess needs to be light |
| 12:35.04 | [TK]D-Fender | tparcina: Actually, that GETS or SETS values for an existing call. |
| 12:35.09 | tparcina | [TK]D-Fender: OK, thank you. I'll try it. |
| 12:35.12 | [TK]D-Fender | tparcina: ORIGINATE is what you're looking for. |
| 12:35.23 | Kobaz | the popular one is kamillio |
| 12:35.51 | cerienjean | right - formerly opensips ? - isnt it using a hammer to kill a fly ? |
| 12:36.09 | Kobaz | it's a sip proxy |
| 12:36.16 | cerienjean | I mean, sophisticated stuff to do a simple task ? |
| 12:36.17 | Kobaz | any sip proxy you use is going to be fairly complex |
| 12:36.25 | cerienjean | ok - thanks |
| 12:36.40 | Kobaz | because people have many different needs for a sip proxy |
| 12:37.05 | tparcina | [TK]D-Fender: I guess the application first calls my phone number, then the other one, and at the end connects those two calls. |
| 12:38.03 | [TK]D-Fender | tparcina: I would not "guess" that. |
| 12:38.08 | tparcina | [TK]D-Fender: You know all those stuff or use some web page I'm not familiar with? |
| 12:38.11 | [TK]D-Fender | tparcina: I would also not "guess" at all. |
| 12:39.11 | tparcina | [TK]D-Fender: In you know all those stuff I ment about GETS, SETS and ORIGINATE. |
| 12:39.36 | [TK]D-Fender | tparcina: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 12:39.42 | tparcina | And yes, it's smart not to assume things. I should know that by now. :) |
| 12:41.23 | [TK]D-Fender | tparcina: "command" is an automatic escalation to "practically take over the entire server" |
| 12:42.04 | cerienjean | kobaz: so asterisk binds to one NIC, the sip proxy binds to the other - I figure out easily the incoming part, but how asterisk should reply so the proxy is used ? assuming client is X -> Proxy Public IP -> Proxy -> Asterisk - should the proxy modify From/To ? How would asterisk reply ? |
| 12:42.32 | [TK]D-Fender | cerienjean: to the proxy of course |
| 12:43.12 | cerienjean | agreed - then how does the proxy knows where to forward the packet ? |
| 12:43.16 | cerienjean | message |
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| 12:43.23 | Kobaz | based on how you configure it |
| 12:43.59 | [TK]D-Fender | cerienjean: because SIP ROUTE headers tell it.... |
| 12:44.10 | [TK]D-Fender | cerienjean: Go read up on how SIP works. |
| 12:44.31 | cerienjean | ok - will do and revert if need be - thanks |
| 12:44.44 | [TK]D-Fender | cerienjean: You'll also need one that proxies the MEDIA as well, not just the signalling |
| 12:44.54 | Kobaz | actually, a little overkill but you could run asterisk on two different interfaces as proxies |
| 12:45.15 | WIMPy | Time to upgrade to 3.9.0. CVE-2013-2094 |
| 12:45.19 | Kobaz | maybe with reinvites, and then have a main asterisk actual media server |
| 12:45.22 | [TK]D-Fender | Kobaz: Depends what you intend to do with it, but that could be possible. |
| 12:45.42 | cerienjean | ok - thanks for the ideas |
| 12:45.43 | [TK]D-Fender | Kobaz: reinvite = DOA |
| 12:45.59 | Kobaz | oh right |
| 12:46.02 | [TK]D-Fender | Kobaz: because it'd try to hop IP to the one the main is bound to. |
| 12:46.05 | Kobaz | you'll need to proxy the call entirely |
| 12:46.12 | Kobaz | yeah exactly |
| 12:46.17 | Kobaz | which is the original problem trying to be solved |
| 12:46.42 | Kobaz | that's why i said maybe |
| 12:46.43 | Kobaz | heh |
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| 12:57.01 | Kobaz | do de do de do breaking stuff |
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| 13:26.19 | toresbe | ChannelZ: thanks for the advice :) |
| 13:26.52 | toresbe | ChannelZ: I'll probably just buy a Digium card to play around anyway, so I'll test it on my own kit. |
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| 13:40.37 | Katty | leifmadsen: congrats on your collaborative effort going into print |
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| 13:51.30 | leifmadsen | Katty: w00t |
| 14:10.23 | zpotoloom | CVE-2013-2094 |
| 14:11.26 | zpotoloom | hmm, that definately wasn't ctrl+c :P |
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| 14:24.05 | msaraiva | Is there too much of a performance penalty by compiling Asterisk with MALLOC_DEBUG enabled? |
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| 14:24.22 | leifmadsen | msaraiva: there can be a penalty on that, yes |
| 14:24.25 | pabelanger | define too much |
| 14:25.48 | msaraiva | Humm, thanks for the prompt answer. |
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| 14:27.02 | msaraiva | I've seem a spike in memory consumption going from Asterisk 1.8 to Asterisk 11. |
| 14:27.17 | msaraiva | So i was looking into tracking down the reason. |
| 14:28.01 | msaraiva | *seen |
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| 15:11.55 | igcewieling | sleep (n): that thing between "I don't understand this at all!" at 3am and "Oh, that was easy" at 9am. |
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| 15:34.59 | emamdouh | hi, how to create an extension for a fax device ? |
| 15:36.03 | leifmadsen | exten => fax,1,NoOp() |
| 15:39.13 | emamdouh | leifmadsen: sorry ? , I'm using trixbox |
| 15:39.26 | leifmadsen | oh, then you should use #freepbx channel |
| 15:39.33 | leifmadsen | this is for vanilla asterisk |
| 15:39.50 | leifmadsen | I've used that gui interface exactly 1 time |
| 15:40.41 | emamdouh | ok thanks |
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| 15:55.19 | MrNemo | ребят русские есть? |
| 15:55.32 | Qwell | If anybody gets messages from Dan French / caterwaul, please send me a copy. Hopkins PD is interested in seeing them. |
| 15:56.03 | Greenlight | Oooh sounds like that ones excalated since last time ^^ |
| 15:57.11 | sweeper | wut |
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| 15:57.16 | sweeper | voip drama |
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| 16:26.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:29.57 | *** join/#asterisk blackbag (~blackbag@gateway/tor-sasl/blackbag) |
| 16:34.19 | MrNemo | есть кто по-русски говорит? |
| 16:34.49 | navaismo | 00 |
| 16:34.50 | Greenlight | MrNemo: This is generally an English-speaking channel, if you can ask your question in English, you may have a better response |
| 16:36.46 | MrNemo | ok, sorry please |
| 16:37.08 | *** join/#asterisk protocolus (ad4ac54c@gateway/web/freenode/ip.173.74.197.76) |
| 16:37.15 | protocolus | hi all |
| 16:37.53 | protocolus | anyone know how to save the audio from SayUnixTime? |
| 16:38.14 | protocolus | i want to play it back with a backgroundDetect |
| 16:42.42 | *** join/#asterisk Gaiax (~Ing-Sopor@unaffiliated/gaiax) |
| 16:42.54 | protocolus | welcome Gaiax |
| 16:43.06 | Gaiax | protocolus: thankx bro. |
| 16:43.32 | protocolus | you know a lot about asterisk? |
| 16:43.49 | Gaiax | lol no man.. |
| 16:44.04 | Gaiax | just a little bit.. i'd love to. |
| 16:44.09 | protocolus | i hear ya me too |
| 16:44.39 | Gaiax | i use more elastix but, im reading about asterisk |
| 16:45.36 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:46.00 | protocolus | i'm trying to figure out how to save the output of the agi SayUnixTime to a file |
| 16:46.15 | protocolus | so i can play it back with background detect |
| 16:46.26 | navaismo | have you tried with monitor/mixmonitor |
| 16:47.00 | protocolus | i could use monitor but i only know how to do tht while on a call |
| 16:47.25 | protocolus | i'm trying to leave a voicemail |
| 16:47.38 | protocolus | and i restart the message on backgrounddetect |
| 16:47.53 | navaismo | the same thing just add the monitor/mixmonitor before dial and check if the audio is saved |
| 16:48.15 | protocolus | but if the beep happens right when i'm doing sayunixtime it doesn't work because it doesn't detect |
| 16:48.25 | protocolus | ohh |
| 16:48.30 | navaismo | anyway i dont fully understand your request, just saying which apps can help you. |
| 16:48.35 | protocolus | i have a channel before dial |
| 16:49.00 | protocolus | no thank you i think i see what you mean |
| 16:52.27 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:54.51 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:55.12 | protocolus | navaismo: so i am trying to make outgoing calls using call files and leave voicemails to clients about upcoming appointments, can i make a call file that has a "local" channel so i can record the wav from the sayunixtime app |
| 16:55.19 | Katty | ASTERISK, etc. |
| 16:55.26 | protocolus | then i could make the real call file to the client |
| 16:55.29 | Katty | et cetera. |
| 16:56.54 | protocolus | hi Katty |
| 16:58.08 | navaismo | protocolus, yes you can create call files pointing to a local channel, and there in the context you can control whatever you want |
| 16:58.42 | protocolus | what is that call i only ever did IAX, SIP DAHDI etc |
| 16:58.53 | Katty | protocolus: ohai. |
| 16:59.48 | navaismo | instead using sip/number you use local/exten@context/n |
| 17:00.08 | protocolus | hehe easy stuff always looks hard |
| 17:00.30 | Katty | i love call files. |
| 17:00.35 | protocolus | me too! |
| 17:01.04 | Katty | you can use a geovision server to tell you when someone paces a set line in a video camera. |
| 17:01.07 | Katty | that kicks off an email. |
| 17:01.17 | Katty | that email can then turned into a call file. |
| 17:01.27 | Katty | with some parsing, and/or cron job stuffs. |
| 17:01.34 | Katty | which can then go RINGING |
| 17:01.37 | Katty | UPS is here. |
| 17:02.36 | protocolus | i want to do a cheap overhead paging system what's a good way to do it? |
| 17:02.42 | Katty | apollo. |
| 17:03.28 | Katty | since you can pipe commands into it, you can issue bash from dialing extensions |
| 17:03.31 | Katty | and make it do stuff |
| 17:03.35 | Katty | with open ssh keys. |
| 17:03.39 | protocolus | nice |
| 17:03.44 | protocolus | you have a link? |
| 17:03.54 | navaismo | protocolus, use the audio from the server and connect many speakers to it :) |
| 17:04.02 | Katty | ^- using apollo. |
| 17:04.25 | _Corey_ | protocolus: Buy an ATA and one of the analog paging amps from Bogon or Valcom... probably the most traditional solution (+/- $300) |
| 17:04.51 | protocolus | sounds easy _Corey_ |
| 17:05.03 | Katty | but piping commads is so much more fun :< |
| 17:05.32 | protocolus | true that Katty |
| 17:05.34 | Katty | someone triggers an event on the vid. surv system, which sends a notification to the phone system, which sends a notification to pipe audio the another system to the speakers |
| 17:05.47 | Katty | UPS IS HERE |
| 17:07.02 | _Corey_ | protocolus: Simple is safe |
| 17:07.09 | protocolus | oh they have sip enabled ceiling speakers with POE |
| 17:07.46 | _Corey_ | Cyberdata has that stuff... it's more expensive though |
| 17:09.45 | protocolus | yeah that was what i was looking at |
| 17:11.02 | protocolus | Katty what kind of video system are you using? |
| 17:11.16 | *** join/#asterisk jkister (~chatzilla@67.200.119.94) |
| 17:13.37 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:16.23 | jkister | someone please ban caterwaul for spam |
| 17:19.31 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:19.31 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 17:25.19 | navaismo | hehe sharing the video? |
| 17:25.27 | Katty | protocolus: anything with email alerts would probably work |
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| 17:27.55 | [TK]D-Fender | jkister: Not what it was for, and not the way to ask. Also extremely unlikely, especially getting the request from some random user. |
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| 17:36.45 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 17:36.47 | saint_ | hi all |
| 17:40.30 | navaismo | o/ |
| 17:40.53 | saint_ | does Digium offer 1 G729 for free, or no ? |
| 17:41.06 | saint_ | I thought I read that somewhere that the 1st one was free.. |
| 17:41.12 | navaismo | as far i know no |
| 17:41.33 | navaismo | only hpecs license with the registration of the digium card |
| 17:41.40 | saint_ | ok, thanks |
| 17:41.44 | navaismo | or Free fax for asterisk |
| 17:41.58 | saint_ | I'm using VOIP.MS as an add on with google voice. |
| 17:42.06 | saint_ | I wish I could find a provider that would support SRTP |
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| 17:44.27 | *** mode/#asterisk [-b *!*@c-174-53-131-243.hsd1.mn.comcast.net] by Qwell |
| 17:44.28 | *** mode/#asterisk [+b *!*@*.hsd1.mn.comcast.net] by Qwell |
| 17:44.30 | *** kick/#asterisk [Sheebacatfood!~north@pdpc/sponsor/digium/Qwell] by Qwell (Sheebacatfood) |
| 17:44.38 | saint_ | who the heck is caterwaul |
| 17:45.17 | _Corey_ | saint_: Someone sending random PMs |
| 17:45.43 | saint_ | yeah. he is telling me "hey , check my youtube video , i explain how to get g729" .. yeah right - |
| 17:46.56 | pabelanger | If caterwaul is harassing you, hope over to #freenode and complain |
| 17:47.08 | pabelanger | I believe he already got a warning or something |
| 17:50.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 17:53.37 | saint_ | I added him in my ignore list .. |
| 17:53.42 | saint_ | thanks for the info though |
| 17:55.13 | saint_ | can someone explain to me the context=xxx in the [general] section of sip.conf .. |
| 17:55.21 | saint_ | how is this used in the whole picture ? |
| 17:55.25 | saint_ | is it just to give a name ? |
| 17:55.42 | saint_ | I have a register = xxxxx |
| 17:56.14 | saint_ | or does it mean i need to have the same context in my extension.conf for incoming calls ? |
| 18:13.09 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 18:13.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:27.29 | [TK]D-Fender | saint_I added him in my ignore list .. < ? |
| 18:27.37 | [TK]D-Fender | What'd I miss? |
| 18:28.52 | navaismo | a spam mesasge with video tutorial |
| 18:29.47 | *** join/#asterisk youjelly (~youjelly@39.47.111.216) |
| 18:30.25 | [TK]D-Fender | Ah yes.... |
| 18:35.37 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 18:38.44 | leifmadsen | navaismo: ya.... |
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| 18:42.18 | tadspoles | h |
| 18:51.44 | *** join/#asterisk mivok (~mark@c-68-55-113-29.hsd1.md.comcast.net) |
| 18:52.21 | mivok | Is it possible on the asterisk cli to evaluate a function and print its result? (Similar to how you can do NoOp(something) to print it in the log? |
| 18:53.05 | igcewieling | mivok: "core show applications like log" |
| 18:53.16 | igcewieling | "core show applications like verbose" |
| 18:55.41 | mivok | I'm not sure that's what I'm looking for - I want a way to run asterisk -r, then see the result of a dialplan function such as MAILBOX_EXISTS(foo) |
| 18:56.02 | mivok | basically to interactively work out what I should be putting in my dialplan without having to actuall make the change/reload each time. |
| 18:56.33 | *** join/#asterisk KNERD (~KNERD@24.175.249.177) |
| 18:57.01 | igcewieling | mivok: that is not something which can be done, however setting higher debug levels might provide similar information |
| 18:57.35 | igcewieling | actually, it is not possible at all to do what you want inside of Asterisk |
| 18:57.52 | mivok | ok, thanks |
| 18:57.53 | [TK]D-Fender | mivok: "voicemail show users" <- |
| 18:58.09 | [TK]D-Fender | mivok: that will show you what boxes exist |
| 18:58.22 | [TK]D-Fender | mivok: But no, you cannot execute dialplan logic from CLI. |
| 19:04.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 19:34.39 | asilva | is anyone looking at this - ASTERISK-21787 ? |
| 19:34.39 | LieutPants | [ASTERISK-21787] [Status: Triage] No IAX2 communication either user/peer or friend accounts - https://issues.asterisk.org/jira/browse/ASTERISK-21787 |
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| 20:02.38 | navaismo | can't find translation for triage is there a synonymous? |
| 20:02.40 | cusco | someone is having pong problems |
| 20:03.35 | ChannelZ | My paddle controller broke. |
| 20:05.39 | asteriskandy | oh lol sorry. i am rebooting trying to change ip. testing phone behavior when net goes down |
| 20:07.14 | wdoekes | navaismo: triage = determining the priority |
| 20:07.54 | wdoekes | bugs are in triage when submitted and not "accepted" and prioritized yet |
| 20:08.30 | navaismo | oh, thanks for the info wdoekes |
| 20:34.27 | sweeper | maaaan. I hate this. I'm working on an interctive classroom environment that uses webrtc, and the management types chose 'ICE' as the abbreviation. |
| 20:34.42 | sweeper | I can't see it used in that context without twitching |
| 20:35.46 | navaismo | uh? |
| 20:36.47 | sweeper | navaismo: http://en.wikipedia.org/wiki/Interactive_Connectivity_Establishment <-- this is the more common use of the acronym. and it's something I use every day in code to make things work behind NAT |
| 20:36.59 | sweeper | so the cognitive dissonance is significant |
| 20:37.36 | navaismo | o..ok; still lost |
| 20:41.29 | sweeper | navaismo: the thing we internally call "ICE" is very different from the thing I've used for years and am using now that is called "ICE" |
| 20:42.10 | navaismo | got it |
| 20:49.26 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
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| 21:02.01 | saint_ | is there a recommendation in sip.conf or any other .conf file , about leaving a space on each side of the = signs ? |
| 21:02.29 | saint_ | ie: would "videosupport = yes" work ? or should I use "videosupport=yes" without spaces around the =- sign ? |
| 21:02.51 | file | doesn't matter |
| 21:02.52 | leifmadsen | both should work |
| 21:03.05 | leifmadsen | I don't use spaces, but that's a style preference |
| 21:03.17 | saint_ | i just found out by using sip show settings .. |
| 21:03.21 | saint_ | i can see YES in front of video support. |
| 21:03.22 | saint_ | thanks |
| 21:03.34 | *** join/#asterisk tech_travis (~Travis@174.46.237.98) |
| 21:04.05 | saint_ | what's this about: Sending fake auth rejection for device xxxxx ..? |
| 21:04.11 | saint_ | xxx being a digium phone .. |
| 21:05.18 | [TK]D-Fender | saint_: It's an auth challenge |
| 21:05.51 | saint_ | mmhh.. i might have broke something |
| 21:06.01 | Maliuta | leifmadsen: spaces are generally a no-no. Tabs are the way to go, then any dev can set his tab stop to however many chars he wants |
| 21:06.42 | leifmadsen | Maliuta: we're talking about different things, and tabs are a no-no in ruby and python land |
| 21:07.02 | leifmadsen | environment and context means everything |
| 21:07.16 | navaismo | screw tab and spaces just program by enters XD |
| 21:07.17 | leifmadsen | but in this case, using tabs/spaces is non-relevant |
| 21:08.00 | Maliuta | leifmadsen: you just proved why python and ruby are crap ;) |
| 21:08.12 | *** part/#asterisk jkister (~chatzilla@67.200.119.94) |
| 21:08.15 | *** kick/#asterisk [Maliuta!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (no flaming) |
| 21:08.43 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 21:08.48 | Maliuta | an we should all be programming in raw binary anyway ;) |
| 21:11.59 | Maliuta | I just got to use a reference are to patching on FB as a pun, and someone actually got it |
| 21:18.12 | saint_ | Is there a case sensitivity in the name of peers ? |
| 21:18.29 | saint_ | I'm getting this error: Registration from '"HOME" <sip:office@192.168.1.242>' failed for '192.168.1.117:5060' - No matching peer found |
| 21:18.41 | saint_ | But I named my peer Office (with an uppercase O) in sip.conf |
| 21:19.02 | [TK]D-Fender | saint_: pastebin the peer masking only the secret |
| 21:20.29 | Maliuta | [TK]D-Fender to the rescue! |
| 21:21.15 | saint_ | [TK]D-Fender: http://pastebin.com/LhrWpxyU |
| 21:21.50 | [TK]D-Fender | type=phone <- BAD |
| 21:22.11 | navaismo | Yes case sensitive |
| 21:22.24 | [TK]D-Fender | scratch that |
| 21:22.56 | [TK]D-Fender | [Office](home) <- what is this (home)? |
| 21:22.56 | Maliuta | Wait, when did type=phone become valid? |
| 21:23.08 | [TK]D-Fender | Maliuta: res_digium <- |
| 21:23.26 | Maliuta | [TK]D-Fender: as of release ... ? |
| 21:23.47 | [TK]D-Fender | Maliuta: Dunno, just that it at least wasn't referencing sip.conf :) |
| 21:23.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ykzwwhpuwjkermtw) |
| 21:24.06 | Maliuta | saint_: the (home) should be a comment somewhere |
| 21:24.40 | saint_ | [TK]D-Fender: http://pastebin.com/index/LhrWpxyU , with the home template |
| 21:25.04 | saint_ | ouch, hold on |
| 21:25.12 | navaismo | Office != office |
| 21:26.31 | [TK]D-Fender | BRB |
| 21:26.53 | Maliuta | and [Office](home) seems totally invalid |
| 21:27.37 | navaismo | Maliuta, its a valid template |
| 21:27.42 | saint_ | Here, with everything : http://pastebin.com/iqHbVVm3 |
| 21:27.54 | navaismo | Office inherits the "home" template values |
| 21:27.57 | saint_ | (home) is a template |
| 21:28.01 | saint_ | see this new link http://pastebin.com/iqHbVVm3 |
| 21:28.36 | saint_ | My issue is that the phone "Office" is trying to register with office@xxx (lowercase o) - So my question was: Is the registration case sensitive ? |
| 21:28.49 | saint_ | Because nowhere I have "office" with a lowercase o .. |
| 21:28.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:29.07 | saint_ | [TK]D-Fender: I was saying that this is the new pastebin with the home template: http://pastebin.com/iqHbVVm3 |
| 21:29.10 | navaismo | hate been ignored |
| 21:29.46 | saint_ | [tk]d-fender: My issue is that the phone "Office" is trying to register with office@xxx (lowercase o) - So my question was: Is the registration case sensitive ? |
| 21:30.14 | Maliuta | saint_: you want the quick fix? chage Office to office :) |
| 21:30.28 | saint_ | Maliuta: that is valide, but I would like to understand though :)_ |
| 21:30.47 | navaismo | lalalala hey lalalalal hey hey *dancing around the fire* |
| 21:30.55 | Maliuta | saint_: something somewhere is forcing lower case (similar to what the SMTP RFC's demand) |
| 21:31.04 | Maliuta | saint_: full stop |
| 21:31.16 | navaismo | Office != office |
| 21:31.21 | Maliuta | I'd have to read the SIP RFC's to be sure |
| 21:31.28 | Maliuta | navaismo: we know |
| 21:31.31 | saint_ | navaismo: okay. but nowhere in the conf. I have office |
| 21:31.41 | navaismo | facepalms |
| 21:31.43 | saint_ | I only have Office |
| 21:31.53 | Maliuta | joins navaismo |
| 21:32.43 | Maliuta | saint_: if the RFC forces lower case then the client will take your "Office" and turn it to "office" before transmission |
| 21:33.00 | Maliuta | saint_: which doesn't match [Office] |
| 21:33.15 | Maliuta | so fix the sip.conf |
| 21:33.27 | saint_ | Maliuta: understand that. rfc 3261 ? |
| 21:33.35 | Maliuta | you can't change the RFC or client behaviour |
| 21:33.54 | [TK]D-Fender | saint_: "sip show peer Office" <- |
| 21:34.04 | Maliuta | saint_: I'm not reading an RFC at 07:30 with no sleep |
| 21:34.33 | saint_ | [TK]D-Fender: that works, you want the pastebin ? |
| 21:34.42 | Maliuta | [TK]D-Fender: I would have though just a "sip show peers" |
| 21:34.49 | [TK]D-Fender | saint_: Clearly |
| 21:35.25 | Maliuta | This is making my Earl Grey go cold, and stopping me messing with people on FB |
| 21:35.33 | saint_ | [TK]D-Fender: http://pastebin.com/7rR2Qb5A |
| 21:36.47 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 21:37.06 | saint_ | Ha.. Is this it ? |
| 21:37.10 | saint_ | http://www.apps.ietf.org/rfc/rfc3261.html , page 158 |
| 21:37.14 | saint_ | look for "lower case" |
| 21:37.31 | saint_ | I'll switch Office to office. We'll see. stand by |
| 21:38.36 | saint_ | that worked |
| 21:38.39 | saint_ | lesson learned. |
| 21:38.41 | Maliuta | saint_: effectively that means that yes, lower case only |
| 21:38.49 | Maliuta | saint_: read the RFC first? |
| 21:38.58 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 21:39.36 | malcolmd | nah, something else is at play. i've taken your paste bin and successfully registered my phone using "Office" |
| 21:39.37 | Maliuta | although that said * should be forcing the sip peers to lower case inline with those RFC references |
| 21:40.08 | saint_ | malcolmd: using DPMA ? |
| 21:40.29 | saint_ | let me try with a regular sip phone, without using DPMA |
| 21:41.15 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:41.17 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:41.27 | malcolmd | yup. i copied in your sip.conf stuff (substituting a context i use for dial plan stuffs called "testing") and in your res_digium_phone.conf stuff (substituting my existing all-networks network). |
| 21:41.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.137) |
| 21:41.59 | saint_ | malcolmd: I confirm. a sip phone registering directly can send Office. If I try this with DPMA (at least with 1.8.5cert1), it will not send Office |
| 21:42.32 | malcolmd | it will, i'm doing it. albeit i'm on asterisk 11 branch. |
| 21:42.55 | saint_ | thank you for your help all by the way. |
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| 22:09.32 | saint_ | malcolmd: i had the same issue with 2 other digium phones provisionned through dpma. i have it working with all lowercase, but if you have a chance you might want to have one of your tech guys give it a shot on 1.8.5cert1 |
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