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| 00:38.14 | apb1963 | Greetings asteriskians |
| 00:38.24 | *** part/#asterisk fireman_biff (~biff@65.48.222.203) |
| 00:51.03 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
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| 01:19.50 | Katty | Asterisk! woo! yeah! |
| 01:29.20 | carrar | YEAH BABY |
| 01:39.26 | leifmadsen | Katty: FREESWITCH! |
| 01:39.35 | leifmadsen | hahahaha I'm totally joking. |
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| 01:39.44 | carrar | no your now!! |
| 01:39.46 | carrar | not |
| 01:40.11 | leifmadsen | that still didn't make sense ;) |
| 01:40.22 | leifmadsen | I think you meant "no you're not" |
| 01:41.00 | carrar | I think you might be right |
| 01:41.12 | carrar | <- engrish fail |
| 01:41.38 | leifmadsen | me fail english? that's unpossible! |
| 01:44.14 | apb1963 | sounds perfectly cromulent to me |
| 01:45.03 | apb1963 | but then I'm usually paffled |
| 01:45.25 | apb1963 | ooops... I meant buzzled |
| 01:48.23 | leifmadsen | buzzed? |
| 01:48.26 | leifmadsen | I prefer that |
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| 02:06.04 | igcewieling1 | Sounds like a story on The Onion: The world was shocked today to learn leifmadsen is really a mole for FreeSwitch |
| 02:06.35 | leifmadsen | been so for years |
| 02:06.51 | leifmadsen | in fact, and you can spread this rumor, but i invented and started freeswitch |
| 02:08.17 | igcewieling1 | little know fact, FreeSwitch came from the ancient Sumerian words Freh and Istich, which mean "angry ex asterisk user" |
| 02:18.13 | Maliuta | lol |
| 02:20.34 | slav3_kitten | lol |
| 02:20.38 | slav3_kitten | i like that |
| 02:20.52 | slav3_kitten | fixed the majority of my asterisk issues |
| 02:21.53 | slav3_kitten | wasn't anything on my gear. apparently someone dicked things up at the ISP. setting my MSS on the edge router to anything higher than 576 causes odd isolated issues |
| 02:23.59 | atan | slaps leifmadsen around a little bit with a large freeswitch |
| 02:24.18 | leifmadsen | igcewieling1: you win a cookie |
| 02:25.30 | atan | Are there any examples of things you *can't* do with Asterisk? So far it's been knock on wood for me |
| 02:26.15 | atan | Actually, I lie. I wanted to wire it into a doorbell easily without any fancy equipment. That was odd, but workable. |
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| 02:44.47 | ruben231 | hi guys anyone have idea and help for this somehow ---> http://pastebin.com/Q15mbz4c |
| 02:44.57 | tzanger | tempts leifmadsen with a leafs victory |
| 02:45.13 | ruben231 | using asterisk 1.8 on centos |
| 02:45.27 | mmlj4 | You do not appear to have the sources for the 2.6.21.7-2.ec2.v1.2.fc8xen kernel installed. |
| 02:45.43 | mmlj4 | install the kernel source |
| 02:46.00 | mmlj4 | line 8 |
| 02:50.36 | ruben231 | <PROTECTED> |
| 02:51.02 | mmlj4 | something like that, yes |
| 02:51.04 | atan | yum install kernel-devel |
| 02:51.17 | mmlj4 | your distro may vary |
| 02:53.12 | ruben231 | <PROTECTED> |
| 02:53.17 | ruben231 | after i install dahdi |
| 02:53.24 | mmlj4 | then you haven't installed the kernel source yet |
| 02:54.45 | ruben231 | mmlj4: please help adn guide how to do it |
| 02:55.01 | atan | http://wiki.centos.org/HowTos/I_need_the_Kernel_Source |
| 02:55.13 | mmlj4 | you were just told how |
| 02:55.24 | mmlj4 | twice, now |
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| 03:00.00 | ruben231 | mmlj4: http://pastebin.com/E11rXLLn |
| 03:00.08 | mmlj4 | nyet |
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| 03:39.59 | Rahail | Hi question I am not tecy person just wondering is it possible to make iax2 protocol use tcp insted of UDP |
| 03:55.33 | mmlj4 | what did google tell you? |
| 03:55.44 | Rahail | yes and no |
| 03:55.47 | Rahail | so manyt hing |
| 03:57.00 | mmlj4 | read this: http://www.voip-info.org/wiki/view/IAX |
| 03:57.15 | mmlj4 | IAX2 uses a single UDP port 4569, and thus works well in NAT environments (the obsolete IAX1 protocol used port 5036). IAX2 uses ONLY one udp port for both control and data traffic. |
| 03:57.48 | Rahail | some countires blocking the UDP |
| 03:58.09 | Rahail | this why I was thinking if possible tcp then it would be hard for them |
| 04:10.29 | carrar | put it in IPSEC |
| 04:11.04 | Rahail | IPSEC sorry i am not sure what is that give me little hint plz |
| 04:11.19 | carrar | VPN Tunnel |
| 04:12.22 | Rahail | thank you ... |
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| 04:36.50 | [TK]D-Fender | GRE used by ipsec is also often blocked. For TCP you'd be looking at an SSL type VPN like OpenVPN |
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| 04:39.30 | Rahail | we got spammer |
| 04:39.35 | Rahail | lmao |
| 04:39.48 | Rahail | crying about post getting delet and sending some image link |
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| 06:26.07 | Assid | heya |
| 06:36.53 | Assid | ok.. so i think i have issues with enbloc/overlap |
| 06:40.32 | Assid | i am seeign a dump of the calls on the NEC; it recognises the caller id and the other functionality work . IF i enable enbloc mode. BUT if i do that. I am unable to simulate my dialtone |
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| 09:08.38 | polysics | hi there |
| 09:08.53 | polysics | is there a way to force a SIP peer to only be on one call at a time, ever? |
| 09:08.57 | polysics | including conferences etc |
| 09:09.26 | polysics | and counting inbound and outbound |
| 09:13.08 | polysics | or am I doing it wrong? |
| 09:14.06 | eirirs | yes you can |
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| 09:17.43 | eirirs | polysics: call-limit = 1 |
| 09:18.07 | eirirs | polysics: need to be asterisk 1.8 or newer |
| 09:18.24 | eirirs | type=peer are required for that to work |
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| 09:19.44 | polysics | peer, not friend? |
| 09:19.50 | eirirs | not friend |
| 09:19.54 | eirirs | not user |
| 09:20.10 | polysics | oh, then that is it |
| 09:24.42 | polysics | I wonder if there is a detailed explanation of what peer vs friend vs user means |
| 09:25.26 | polysics | but it does look like we should just use peer |
| 09:25.54 | eirirs | yes, no reason for using friend/user now anymore |
| 09:28.13 | eirirs | caterwaul keep PM'ing me |
| 09:28.37 | polysics | me too |
| 09:28.52 | polysics | I really don't like video tutorials, by the way :-D |
| 09:29.14 | polysics | give me a coherent, SHORT blog post and I am fine |
| 09:29.31 | polysics | I have spent too much time fixing stuff someone cluelessly built from a tutorial :-) |
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| 09:33.08 | Assid | anyone played with other PBX systems ? like siemens ? |
| 09:33.44 | eirirs | polysics: agreed |
| 09:33.59 | eirirs | now he wrote "please don't stonewall me, tanks" |
| 09:34.17 | polysics | Assid: I have I would say very good experience with FreeSWITCH-based PBXs |
| 09:34.38 | Assid | err no.. i meant like them "proprietory" types |
| 09:34.52 | eirirs | I use FreePBX, and before that, Trixbox |
| 09:34.59 | Assid | was wondering how they behave with the likes of asterisk |
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| 09:36.25 | polysics | no experience there, but if they speak SIP properly you can surely get them to do things for you |
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| 09:36.52 | polysics | we have setups where Asterisk is a B2BUA providing call applications to an old system |
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| 09:39.33 | kaldemar | polysics: GROUP functions. call-limit was deprecated in 1.6.0. |
| 09:43.28 | polysics | had to block the guy |
| 09:43.36 | polysics | he is pure nonsense |
| 09:43.40 | polysics | this caterwaul guy |
| 09:44.11 | polysics | kaldemar: I am not even using the dialplan :-) |
| 09:44.35 | eirirs | kaldemar: can you point me to a URL with some examples? |
| 09:44.40 | polysics | +1 |
| 09:45.10 | polysics | I found this, a little amateurish but pretty clear |
| 09:45.12 | polysics | http://www.fruitnotes.com/blogs/Limiting_Calls_on_Asterisk_using_Group_cmd_1255 |
| 09:45.36 | polysics | all this app does is async originates, so no GROUP() stuff |
| 09:45.50 | polysics | why was that deprecated? not everyone does stuff in the dialplan+ |
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| 09:46.42 | polysics | that basically means I have to track extension state by hand, how cool :-/ |
| 09:47.16 | polysics | kaldemar: is busylevel deprecated too? |
| 09:50.00 | kaldemar | eirirs: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Security_id36001703.html |
| 09:50.24 | kaldemar | polysics: no. that requires callcounter=yes to work. group function use is encouraged though. |
| 09:51.00 | polysics | since I am using AMI for everything, I would then have to drop calls in a context with groups instead. |
| 09:51.12 | polysics | I suppose that is still doable |
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| 10:13.22 | polysics | another thing I have been unable to solve: in Confbridge, I have an user sitting waiting for a single customer. When the customer joins, the agent should hear the joining tone, but the customer should not. |
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| 11:59.34 | jkroon | hi guys, when doing Dial(,,t) and the user initiates a transfer it seems to same extension in which the call is when Dial() is executed is used - is there any way to change that to a different extensions? |
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| 12:03.53 | igcewieling1 | jkroon: Are you looking to change the callerid of the transferred call? |
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| 12:15.48 | jkroon | igcewieling1, that's done with connectedline |
| 12:16.13 | igcewieling1 | jkroon: incorrect, but I don't have any more time to help. |
| 12:16.41 | jkroon | no, let's say I'm in context "a" and execute a Dial(SIP/123,,t); then if SIP/123 hits # the dialplan will be entered into the "a" context, i want it to re-enter into a different context, say "b" |
| 12:16.54 | igcewieling1 | to answer your question, you change the extension Dial()'d by dialing different digits when you transfer. |
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| 12:17.44 | igcewieling1 | jkroon: You could do an include => b in the [a] context in your dialplan. |
| 12:17.46 | jkroon | that's fine, but that exten => that you intend to get to must be in the same context as the original Dial() statement ... |
| 12:17.53 | [TK]D-Fender | jkroon: First, why are you using DTMF transfers? |
| 12:17.56 | jkroon | igcewieling1, i'd really prefer to NOT do that. |
| 12:18.16 | igcewieling1 | jkroon: you don't have a choice. you can't magically jump contexts -- contexts are a security measure. |
| 12:18.21 | jkroon | [TK]D-Fender, two reasons: 1. Client with analog phones, 2. Client with a Siemens Gigaset A510IP that *can't* do SIP transfers. |
| 12:18.42 | jkroon | igcewieling1, that's exactly why I don't want to include b into a. |
| 12:19.03 | igcewieling1 | jkroon: then copy the relevant exten line to [a] |
| 12:19.05 | jkroon | because b opens up a lot of routes to the caller, the person doing the transfer is considered "trusted" |
| 12:19.18 | [TK]D-Fender | jkroon: #1 almost every ATA and DAHDI device all support hook-swtich(flash) transfers) |
| 12:20.38 | [TK]D-Fender | jkroon: #2 ${TRANSFER_CONTEXT} <------ |
| 12:20.56 | jkroon | [TK]D-Fender, that looks like what I want - is that in features.conf? |
| 12:21.11 | jkroon | or do I just set that as a channel variable? |
| 12:21.26 | [TK]D-Fender | jkroon: that is clearly a channel variable |
| 12:21.38 | jkroon | channel var :) |
| 12:21.43 | jkroon | cool thanks... |
| 12:22.44 | [TK]D-Fender | And seriously get them using HS transfers, not DTMF |
| 12:22.47 | *** join/#asterisk peektoseen (c169eb01@gateway/web/freenode/ip.193.105.235.1) |
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| 12:24.04 | jkroon | [TK]D-Fender, i've been fighting this so hard it's not even funny. |
| 12:25.00 | peektoseen | hi all. How can I view a version of asterisk? I try 'core show version' and get "Asterisk SVN-trunk-r376131M built by root @ WebAsterisk on a i686 running Linux on 2012-11-16 07:04:12 UTC" . r376131M - it 11 version, or 10, or else? |
| 12:25.22 | [TK]D-Fender | r376131M |
| 12:25.23 | jkroon | lol, it's svn trunk, revision 376131 :p |
| 12:25.31 | [TK]D-Fender | yup |
| 12:25.39 | *** join/#asterisk MrPockets (~MevinKitt@unaffiliated/mrpockets) |
| 12:26.21 | MrPockets | So I'm trying to support this client's Astlinux box (I know, don't say it). Need to forward a phone number into a cellphone. Which config in Asterisk would this generally be in? |
| 12:26.39 | jkroon | extensions.conf ... |
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| 12:28.37 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 12:28.47 | MrPockets | That makes sense.. |
| 12:31.02 | jkroon | [TK]D-Fender, thanks a million. i woudl have searched for a VERY long time before finding that. |
| 12:31.15 | [TK]D-Fender | jkroon: and that seimens DOES have internal transfer features |
| 12:31.27 | [TK]D-Fender | jkroon: Page 58 of their manual |
| 12:31.40 | jkroon | [TK]D-Fender, i'll re-look, i would *highly* prefer that. |
| 12:32.09 | jkroon | if it's the int/r button thing, all that that ends up doing is sending a telephone-event down the sip channel which it expects the PBX to act on. |
| 12:32.27 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 12:32.29 | [TK]D-Fender | jkroon: No, it's with the call & end buttons |
| 12:32.53 | [TK]D-Fender | jkroon: Multple ways. Documentation. It's wonderful. Use it. |
| 12:34.00 | [TK]D-Fender | jkroon: Not the most intuitive thing in the world, but hey, for $90 USD its and SPA-3102 w/ DECT handset. Not bad. |
| 12:34.48 | jkroon | grr, i'm going to slap stephen. |
| 12:36.43 | *** join/#asterisk vfabi (~fabi@host-static-37-75-94-227.moldtelecom.md) |
| 12:43.45 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:44.11 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:48.45 | sweeper | anyone used the asterisk webrtc stuff yet? |
| 12:51.08 | [TK]D-Fender | MrPockets: BTW their docs give a similar warning not to mess with the config files by hand. |
| 12:52.12 | MrPockets | Astlinux? I know. I'm using the web-gui config editor. |
| 12:52.21 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 12:52.32 | MrPockets | Found that out the hard way with some IPTables modification. Thanks for the heads-up though! |
| 12:53.33 | MrPockets | if you don't mind me continuing to bug ya'll. The extensions.conf seems to be the Master dial-plan, dictating how to handle all calls coming in. I want to just take one user's extension and forward it to his cell phone. Shoudl I still keep looking into how to do this with extensions.conf? |
| 12:53.55 | sweeper | MrPockets: yes |
| 12:54.00 | MrPockets | K. Thanks. |
| 12:54.23 | sweeper | MrPockets: all you should have to do is match their extension and then dial the appropriate number |
| 12:54.33 | sweeper | literally one line |
| 12:54.49 | [TK]D-Fender | MrPockets: I'd be betting that this is part that is generated by AstLinux when you apply changes and would get blown away if you attempted to change any of it... |
| 12:54.50 | tenspeed705|work | once you get it, you will be all like ohhhhh |
| 12:55.24 | MrPockets | [TK]D-Fender, that'd be my guess too, but I dont' see any point-and-click GUI to config this |
| 12:55.41 | MrPockets | So i figure I'd start here, and if it over-wrights the config then I'll keep scratch'en my noggin. |
| 12:55.59 | [TK]D-Fender | MrPockets: Chances are WYSIWYG |
| 12:56.05 | *** join/#asterisk serafie (~erin@nat/digium/x-ydrkhrsrhgqsavpf) |
| 13:19.16 | MrPockets | exten => 120,1,Dial(MYCELLNUMBERHERE) |
| 13:19.22 | MrPockets | that look right? (possibly?) |
| 13:21.20 | wdoekes | MrPockets: you'll need a technology and a destination |
| 13:21.33 | wdoekes | e.g. Dial(SIP/mytrunk/MYCELLNUMBERHERE) |
| 13:21.39 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-kafaadvbfowokmdq) |
| 13:21.41 | [TK]D-Fender | !book |
| 13:21.46 | [TK]D-Fender | ~book |
| 13:21.46 | 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 |
| 13:21.48 | [TK]D-Fender | ^^^ |
| 13:23.58 | MrPockets | Good deal. |
| 13:23.59 | MrPockets | thakns. |
| 13:24.12 | MrPockets | (side note: Oreilly has the *MOST RANDOM* shit on the covers of their books |
| 13:24.55 | MrPockets | ACtive Directory 2012: A bear riding a giraffe playing polo with a watermellon on its head. |
| 13:25.07 | [TK]D-Fender | This one is probably the most appropriate of their lineup |
| 13:25.48 | *** join/#asterisk xuw (6e4de5f0@gateway/web/freenode/ip.110.77.229.240) |
| 13:26.43 | xuw | im trying to jump to different context according to the result of a bash script. |
| 13:26.49 | xuw | how would i go about doing that with gotoif? |
| 13:27.42 | xuw | in my asterisk realtime extensions db i currently have this set |
| 13:27.43 | xuw | Set NUMBER=${SHELL(/usr/bin/script.sh 0${EXTEN:2}) |
| 13:28.04 | leifmadsen | seems pretty close |
| 13:28.06 | xuw | wich returns a short text, and i would like to jump to another context if it matches the string |
| 13:28.13 | xuw | which* |
| 13:28.15 | leifmadsen | make sure your script returns a status value |
| 13:28.23 | xuw | it does. i NoOp the value and its correct |
| 13:28.29 | leifmadsen | SHELL() I'm pretty sure just returns the exit value of the script |
| 13:28.37 | leifmadsen | ok, so that's something |
| 13:28.39 | xuw | it returns the value of the script |
| 13:28.41 | leifmadsen | so just setup the GotoIf()? |
| 13:28.50 | xuw | yeah but not really sure how :/ |
| 13:28.56 | leifmadsen | you don't know how to use GotoIf? |
| 13:29.03 | xuw | ive tried and failed :p |
| 13:29.12 | leifmadsen | you should show how you're failing then so people can help |
| 13:29.20 | sweeper | how can I check if I built SRTP into the asterisk I just compiled? |
| 13:29.25 | leifmadsen | it's just: GotoIf($[something = true]?true:false) |
| 13:29.34 | leifmadsen | sweeper: see if res_srtp.so is loaded |
| 13:29.45 | [TK]D-Fender | xuwim trying to jump to different context according to the result of a bash script. <- Just GOTO |
| 13:30.46 | sweeper | leifmadsen: where can I find a current command reference for asterisk 11? |
| 13:30.48 | xuw | so how would i jump to context,s,300 ? |
| 13:30.50 | [TK]D-Fender | Actually... you wanted to compare the result... not that it returns the context name directly. So strike that ... GotoIf indeed... |
| 13:31.04 | [TK]D-Fender | xuw: "core show application GotoIf" <- |
| 13:31.26 | leifmadsen | sweeper: core show help |
| 13:31.35 | xuw | yep ill give it a try again |
| 13:31.40 | leifmadsen | xuw: GotoIf($[....]?context,s,300) |
| 13:31.41 | xuw | thought i was pretty close :p |
| 13:31.46 | leifmadsen | xuw: I think you should really read asteriskdocs.org |
| 13:32.00 | leifmadsen | that's in the Dialplan Basics chapter i'm pretty sure |
| 13:32.06 | xuw | yup. |
| 13:32.06 | xuw | good to have leifer'n around, he knows the tricks. |
| 13:32.16 | sweeper | leifmadsen: excellent |
| 13:32.20 | leifmadsen | there's no trick to it, it's just a simple trick |
| 13:32.32 | xuw | :) |
| 13:33.08 | sweeper | module show lists loaded modules only, correct? |
| 13:33.31 | sweeper | ah, apparently yes :) |
| 13:37.50 | leifmadsen | or 'module show like srtp' |
| 13:41.10 | sweeper | whoooooooo |
| 13:41.40 | sweeper | that was suprisingly easy, just logged into asterisk with webrtc + jsSIP |
| 13:42.00 | sweeper | anyone know if there's a timeline for supporting vp8 passthrough? |
| 13:43.41 | [TK]D-Fender | sweeper: If there is a standard for the offering I don't see why it would take any real amount of work.... * never transcodes video anyway... |
| 13:46.27 | sweeper | I would love to do the whole stack in asterisk instead of handling video separately |
| 13:48.25 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ehehzkeecrgmwkxq) |
| 13:48.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:51.10 | sweeper | [TK]D-Fender: interesting. I'll have to look into adding that then, after I get the current iteration done. I've got a freaking python/gstreamer app handling video atm :P |
| 13:53.39 | sweeper | ok then so now to integrate asterisk with the rest of the application.... |
| 13:55.03 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:01.30 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 14:02.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:02.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.42 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 14:04.56 | sweeper | aside from sip.conf and RealTime, what other options are there for managing sip peer entris? |
| 14:04.59 | sweeper | *entries |
| 14:05.09 | jmetro | realtime and realtime. |
| 14:06.12 | sweeper | hmm. I need something event-y. I could probably make realtime work, but adding a database for the sole purpose of communicating with * is...icky |
| 14:06.28 | jmetro | you do a lot of things with that database |
| 14:06.38 | jmetro | its a very nice database to have. |
| 14:06.47 | Katty | Astersisk! woo! yeah! |
| 14:07.13 | sweeper | yea but pretty much all I need to do is add/remove sip peers. they don't even need to stick around for very long |
| 14:07.16 | Kobaz | axeterisk! |
| 14:07.28 | leifmadsen | sweeper: that's about the only interface for that kind of thing |
| 14:07.39 | leifmadsen | unless you want to do a lot of sip reloads |
| 14:07.41 | *** join/#asterisk blee (~blee@142.196.144.201) |
| 14:07.49 | leifmadsen | sweeper: you could try realtime via curl |
| 14:07.52 | Kobaz | Katty: sometimes i think i should have picked home renovations instead of software development |
| 14:08.06 | Kobaz | Katty: houses don't randomly blow up after you stop looking at them |
| 14:08.24 | leifmadsen | Kobaz: unless you left the propane on |
| 14:08.28 | Kobaz | true |
| 14:08.40 | Kobaz | i just spent 15 hours putting in a bathroom exhaust fan |
| 14:08.49 | sweeper | leifmadsen: is there a realtime driver that lets you execute shell scripts? :D |
| 14:08.50 | leifmadsen | that seems like too long :) |
| 14:08.56 | leifmadsen | sweeper: look at extconfig.conf |
| 14:08.59 | leifmadsen | but no |
| 14:09.02 | leifmadsen | well actually, that's not true |
| 14:09.03 | Kobaz | it does.. but it's a lot of work |
| 14:09.11 | leifmadsen | actually, yes it's true -- in terms of realtime |
| 14:09.21 | leifmadsen | you can use #exec in a sip.conf file,b ut that requires sip reload |
| 14:09.24 | sweeper | depends on how many holes you had to cut and then make look nice |
| 14:10.59 | Kobaz | leifmadsen: cut a pilot hole in the ceiling, realized i needed to go four more inches over for the real opening. cut the hole, made mounting brackets for the fan, mounted the fan, cut a hole for the toggle switch, fished electric lines to the attic from the switch... realized i should do a new circuit so i ran 30 feet of new line from the breaker to the bathroom |
| 14:11.33 | sweeper | a new circuit for an exhaust fan? seems excessive |
| 14:11.37 | leifmadsen | ya, the new breaker was the killer :) |
| 14:11.41 | leifmadsen | +1 on excessive |
| 14:11.43 | Kobaz | leifmadsen: had to fish that line up to the attic and then to the switch box, had to wire up all the switches and then the fan itself... then had to cut a hole in the roof and align all the vent pipes, and then put in the exhaust port in the roof and seal it |
| 14:12.12 | sweeper | open a window \o |
| 14:12.13 | leifmadsen | great, I'll have you over when I need an exhaust fan since you'll be able to do it in half the time now |
| 14:12.31 | Kobaz | well the gcfi in the bathroom that i was going to gang off of, was on the same citcuit as the washing machine... so i wasn't going to put a light switch and the exhaust fan on that too |
| 14:12.54 | Kobaz | sweeper: it's a common misconception that a window alone is enough ventilation for a shower |
| 14:13.21 | Kobaz | sweeper: that's why the paint starts peeling and you get mold buildup on the walls if you don't vent properly |
| 14:13.22 | sweeper | oh for a shower no |
| 14:13.29 | sweeper | poop smell, yea |
| 14:13.35 | Kobaz | poop. yes |
| 14:13.36 | Kobaz | haha |
| 14:14.05 | *** join/#asterisk youjelly (~youjelly@39.47.204.36) |
| 14:14.30 | jmetro | who needs shower exhaust, its good to build up the steeam |
| 14:14.39 | sweeper | man realtime seems like a great abstraction, boggles me why there aren't any generic drivers |
| 14:15.06 | Kobaz | jmetro: sure if you like mold |
| 14:15.32 | jmetro | Kobaz: I dont like mold, but i do like mold-killing paints, metal, and stone. |
| 14:16.21 | Kobaz | mm |
| 14:17.13 | Kobaz | i want to redo my bathroom at some point |
| 14:17.17 | Kobaz | stone sounds cool |
| 14:18.20 | jmetro | i like stone because it stays cold in the heat. and the midwest is @#$ing hot |
| 14:18.39 | sweeper | yea but WINTER IS COMING |
| 14:18.42 | Kobaz | yeah |
| 14:18.46 | jmetro | sweaters bro |
| 14:18.55 | sweeper | sweatrs on your feet? |
| 14:18.56 | Kobaz | wait, wasn't winter just like, a week ago |
| 14:19.12 | jmetro | sweeper: you heard of socks? |
| 14:19.25 | Kobaz | and ugg boots |
| 14:19.33 | jmetro | Kobaz: hell no |
| 14:19.34 | sweeper | jmetro: yea but I'm much more likely to be barefoot in the bathroom |
| 14:19.39 | Kobaz | yuppy girl boots |
| 14:20.06 | jmetro | sweeper: it gets so hot inside the house during winter that i like the cold stone |
| 14:20.56 | sweeper | well looks like I'm adding "generic realtime driver" to my do-after-project-is-working list |
| 14:21.35 | jmetro | copy-paste all the specific ones =) |
| 14:21.37 | sweeper | wonder what the best way to provide that would be...HTTP/JSON or something similar? |
| 14:21.48 | sweeper | yea will probably start with that |
| 14:22.08 | jmetro | I dont like JSON because it sounds like someones name |
| 14:22.27 | sweeper | well it's about as generic as you can get these days |
| 14:22.46 | sweeper | although plaintext might not be terrible, JSON is structured at least |
| 14:23.27 | coppice | I worry that JSON will go crazy on Friday the 13th |
| 14:24.04 | sweeper | any suggestions for a more useful/generic protocol than HTTP? |
| 14:24.04 | *** join/#asterisk eslam (~emamdouh@41.233.209.118) |
| 14:24.09 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 14:24.54 | eslam | i need to list all land lines attacked to asterisk box through asterisk consle |
| 14:25.02 | eslam | can anyone help me with that ? |
| 14:25.23 | sweeper | eslam: wikileaks? |
| 14:26.03 | [TK]D-Fender | eslam: .. "attacked"? |
| 14:26.22 | [TK]D-Fender | eslam: Could you clarify all of that... |
| 14:26.24 | sweeper | he does mean attached :P |
| 14:26.25 | jmetro | i think he meant attached |
| 14:26.31 | jmetro | like plugged in |
| 14:26.50 | eslam | sorry yeah i mean attached not attacked , sorry for that |
| 14:26.52 | sweeper | had a hard time finding even a half-decent joke for that one |
| 14:27.26 | sweeper | eslam: do you want the phone numbers or the FXO devices or what? |
| 14:28.00 | [TK]D-Fender | eslam: What kind of "lines"? |
| 14:28.15 | [TK]D-Fender | eslam: Your terminology is too vague as to what you have on it now... |
| 14:28.17 | eslam | sweeper, i need to make sure that i have 2 land lines attached to asterisk box from asterisk console |
| 14:28.29 | *** join/#asterisk slackytude (1000@37.81.24.68) |
| 14:28.30 | [TK]D-Fender | eslam: What kind of lines? Plugged in how? |
| 14:29.39 | sweeper | eslam: ok if you have hard lines, they must be plugged into some sort of card or addon device. what you really need to check is if that card/device's drivers are loaded and the channel is seen by asterisk |
| 14:30.25 | jmetro | i plug my headset directly into the asterisk box, why cant i make calls through the keyboard ? |
| 14:30.51 | eslam | <[TK]D-Fender>: they are plugged into an analog card |
| 14:31.07 | [TK]D-Fender | eslam: which? |
| 14:31.10 | eslam | sweeper: yeah |
| 14:31.17 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 14:31.41 | sweeper | eslam: so what kind of card is important, since it will determine the drivers/module that need to be checed |
| 14:31.44 | sweeper | *checked |
| 14:31.46 | gnudna | hi guys i just moved from asterisk 1.6 to 1.8 from debian and now my sounds files aka menu promopts do not work |
| 14:32.00 | eslam | [TK]D-Fender: i'm not sure, is there any way from asterisk console |
| 14:32.04 | gnudna | ast_openstream_full: File menu1/genmenu does not exist in any format |
| 14:32.08 | eslam | to get such data |
| 14:32.18 | sweeper | eslam: try lspci from linux shell |
| 14:32.21 | gnudna | im probably missing something obvious |
| 14:32.25 | sweeper | see if you can find it there |
| 14:32.50 | eslam | sweeper: Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 14:33.03 | [TK]D-Fender | eslam: "dahdi show status" should tell you what ports you configured and if the line is detected as plugged in on it |
| 14:33.05 | sweeper | ah there you go |
| 14:33.13 | [TK]D-Fender | eslam: lFor those kinds of cards. |
| 14:33.21 | gnudna | anybody able to assist me quickly |
| 14:33.44 | gnudna | example using debian is there a particular module that needs to be installed for wav or gsm playback? |
| 14:33.45 | sweeper | gnudna: probably the path changed |
| 14:34.00 | gnudna | it looks the same sadly |
| 14:34.01 | [TK]D-Fender | [10:32]gnudnaim probably missing something obvious <- Yes. The file. |
| 14:34.09 | gnudna | no the file is there |
| 14:34.18 | sweeper | gnudna: no, I mean the path that asterisk is looking for the file in |
| 14:34.21 | gnudna | just i do not see where the path is defined |
| 14:34.32 | [TK]D-Fender | gnudna: Wrong permissions or it isn't looking where you think it is and you should be checking "core show settings" |
| 14:34.48 | [TK]D-Fender | gnudna: and the paths are defined in asterisk.conf |
| 14:35.02 | eslam | [TK]D-Fender: Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 14:35.03 | eslam | wrtdm Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 14:35.17 | gnudna | /var/lib/asterisk is all i have |
| 14:35.21 | eslam | i guess this means that i just have one line plugged in, right ? |
| 14:35.23 | gnudna | sounds files are in there |
| 14:35.57 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 14:36.01 | Assid | heya |
| 14:36.10 | gnudna | i think i found it |
| 14:36.12 | gnudna | :( |
| 14:37.25 | eslam | [TK]D-Fender: i paste dahdi show status here for more visability http://pastebin.com/t5CKxMEz |
| 14:37.32 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:37.32 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:37.35 | gnudna | awesome that was it ;) |
| 14:37.54 | [TK]D-Fender | gnudna: "dahdi show channels" might give more. |
| 14:38.04 | eslam | [TK]D-Fender: it seems i have just one land line plugged into sangoma card, right ? |
| 14:38.07 | [TK]D-Fender | gnudna: that single-line one isn't of much use for a multi-port card |
| 14:38.49 | Assid | does it make a difference on the bpviol value ? |
| 14:40.51 | Assid | mins between 539 and 757 on my 2 ports |
| 14:42.45 | *** join/#asterisk igcewieling (~igcewieli@wsip-98-174-63-41.pn.at.cox.net) |
| 14:42.59 | gnudna | no port card straight up sip here |
| 14:43.31 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:43.36 | [TK]D-Fender | gnudna: Sorry, cross-talk error |
| 14:45.21 | *** join/#asterisk Matthias (~Matthias@bnc1.ip6.dirlinger.at) |
| 14:45.22 | *** join/#asterisk Captain_Proton (~quassel@173.162.32.1) |
| 14:46.44 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
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| 14:51.54 | *** join/#asterisk maximilianoo (~maximilia@187.60.66.11) |
| 14:52.49 | evilsk4ter | evil_gordita =) |
| 14:53.23 | Captain_Proton | Anyone know of a good way to fix this problem https://issues.asterisk.org/jira/browse/ASTERISK-5024. I have set the dtmftimeout=1000 it helps but still to many are failing |
| 14:53.42 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.114) |
| 14:55.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:55.32 | igcewieling | Captain_Proton: looks like a patch was committed in 2008. Upgrade |
| 14:55.51 | igcewieling | Polycoms have the option for DTMF and Silence length settings |
| 14:56.38 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 15:01.04 | Assid | if the line coding between 2 devices is wrong.. would the devices connect ? |
| 15:03.53 | jmetro | I feel like that question answers itself |
| 15:05.13 | igcewieling | jmetro: it does answer itself. |
| 15:11.54 | jmetro | \o/ i like being right |
| 15:17.31 | Assid | jmetro: im getting lots of bpviol .. was wondering why |
| 15:17.40 | italorossi | Is there any way to update a member status (unpause/paused) after pausing/unpausing it using realtime members? (queue show queuename is not an option) ast 1.8 |
| 15:18.27 | italorossi | The problem is when there is only one call waiting on queue and the only available member is paused, after unpausing it the call does not get forwarded to the member |
| 15:18.53 | igcewieling | Assid: either a mismatch between the two ends or a circuit problem |
| 15:19.10 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 15:19.10 | igcewieling | Assid: most T-1s are set for ESF/B8ZS |
| 15:19.29 | Assid | ok so heres the progress so far.. i can get my NEC to speak to my asterisk box.. calls come and go.. so far thats fine.. features of the NEC however dont work perfect IF i use overlap mode |
| 15:19.56 | Assid | if i enable overlapdialling and immediate in dahdi.. then my NEC does wonky stuff |
| 15:20.57 | WIMPy | Or is it your dialplan? |
| 15:22.17 | Assid | WIMPy: 02168 (which technically routes the call back to 168 on my nec+ring my sip phone for example.) then it calls some random extension which is IN the nec itself.. |
| 15:22.26 | igcewieling | immediate=yes tells asterisk when the analog port goes off hook to immediatly send the call to the "s" extension. |
| 15:22.46 | Assid | when i disable immediate and overlapdial the extension within NEC worked fine |
| 15:23.00 | WIMPy | still suspects the dialplan. |
| 15:23.17 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 15:23.25 | WIMPy | igcewieling: Analog is evil. |
| 15:23.26 | igcewieling | Assid: good, that is expected. |
| 15:23.27 | Assid | WIMPy: it magically calls the right extension IF thats removed? |
| 15:23.34 | igcewieling | WIMPy: analog is just misunderstood |
| 15:23.41 | Assid | igcewieling: it is ? |
| 15:23.54 | igcewieling | you seldom want immediate=yes and overlap dialing seldom works as you expect |
| 15:24.06 | Assid | my NEC lets me do enbloc mode if i want it to |
| 15:24.07 | igcewieling | using either is ususual |
| 15:24.18 | igcewieling | Assid: overlap dialing is not enloc dialing |
| 15:24.23 | WIMPy | You always wat it, but you do indeed have to be a little carefull to make it work. |
| 15:24.45 | Assid | yes.. thats why im mentioning the option exists.. should i rather use enbloc ? |
| 15:25.10 | igcewieling | Assid: I recommend always using enbloc |
| 15:25.26 | WIMPy | Whatever you prefer, I wouldn;t use enblock. |
| 15:25.35 | Assid | if i enable enbloc.. the call conference and other things work.. then i think i need to tweak the dialplan.. cause my simulated dialtone doesnt work |
| 15:26.18 | Assid | and from what i read.. enbloc causes some frame corruption.. or something along those lines |
| 15:26.24 | gnudna | [TK]D-Fender, thanks for the reference before datafiles is where it was expecting the audio files to be in |
| 15:26.31 | Assid | see 1 person giving me opposite "suggestion" than the other |
| 15:26.34 | WIMPy | What? |
| 15:26.52 | igcewieling | Assid: Um, 95% or more people use enbloc instead of overlap |
| 15:27.27 | igcewieling | Assid: what is the dialplan on the NEC? do you have variable length overlapping patterins? |
| 15:27.33 | WIMPy | Just because you disable it on the Asterisk side doesn;t mean you NEC won;t still do overlap. Dahdi just hides it away from your dialplan. |
| 15:27.55 | Assid | WIMPy: i can change the NEC side to enbloc as well |
| 15:28.11 | igcewieling | Assid: exactly how are you interfacing Asterisk with your NEC? |
| 15:28.18 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.51.221) |
| 15:28.47 | Assid | igcewieling: over pri |
| 15:29.16 | igcewieling | Assid: what is the dialplan on the NEC? do you have variable length overlapping patterins? |
| 15:29.52 | Assid | on the NEC no.. 3 digit.. with 1XX => extensions.. and 0 -> call through pri |
| 15:30.12 | Assid | so when anyone hits 0; i simulate a dialtone.. and they call out |
| 15:30.14 | Captain_Proton | igcewieling: I do have the patch 1.8.16 or somethng the problem is it not working reliably. Is there anything you can think of to help? |
| 15:30.48 | igcewieling | Captain_Proton: I already gave you a suggestion. UPDATE |
| 15:30.55 | WIMPy | Assid: What do you mean by 'simulate dialtone'? A WaitExten should be all you need. |
| 15:30.56 | igcewieling | at a minimum update to the latest 1.8.x |
| 15:31.26 | Assid | exten => s,1,WaitExten(15,m(dialtone)) |
| 15:31.39 | igcewieling | Assid: unless you have variable length overlapping patters, overlapdialing does nothing useful except generate extra traffic on the d-channel |
| 15:32.18 | igcewieling | Assid: why not exten => _XXXX,1,Dial(SIP/${EXTEN}) or similar when you don't need immediate yes |
| 15:32.24 | WIMPy | No need for that m option. |
| 15:32.32 | igcewieling | you have a PRI there is NO NEED for immediate=yes |
| 15:33.06 | igcewieling | your NEC should send the entire dialed number as part of the call setup message. |
| 15:33.32 | WIMPy | But you have to wait for a timeout. |
| 15:33.57 | Assid | yeah that takes a while.. and lots of people including me have analog phones.. like cordless etc |
| 15:34.19 | WIMPy | Analog cordless? Ouch. |
| 15:34.20 | Assid | http://pastebin.com/77W777Xn |
| 15:34.23 | jmetro | strange i have a cordless thats SIP |
| 15:34.51 | Assid | yeah well.. in my case .. i got a cordless + my personal cell phone with sip.. |
| 15:35.10 | Assid | but most people use that crappy 10-20$ handsets.. cause they keep breaking it |
| 15:35.39 | Assid | some people who need to roam around have a panasonic cordless phone |
| 15:35.46 | Assid | anywyas.. so thats the reason i need this.. |
| 15:35.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.168) |
| 15:36.14 | jmetro | what about a PAP2t |
| 15:36.23 | igcewieling | Assid: You have analog phones connected to Asterisk? |
| 15:36.36 | Assid | they connected to the NEC |
| 15:36.46 | Assid | what i have noticed..if i use enbloc.. all the services of the NEC/digital extension works fine.. including call conferencing; redial etc.. |
| 15:36.46 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 15:36.49 | igcewieling | then as far as asterisk is concerned they are PRI |
| 15:36.58 | WIMPy | prefers CAT-iq |
| 15:37.00 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.133) |
| 15:37.16 | Assid | yes... <telco> <--> asterisk <--> <nec> |
| 15:37.30 | Assid | thats how it flows |
| 15:37.55 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.168) |
| 15:38.04 | Assid | if i use enbloc.. then my dialtone doesnt come up |
| 15:38.17 | Assid | if i fix that. i think eveyrtthing works |
| 15:38.25 | igcewieling | Your NEC should collect digits from the analog phones based on the NEC dialplan settings, then it should send all the digits at once to Asterisk as part of the call setup message. This is called ENBLOC ialing |
| 15:39.01 | igcewieling | Assid: Why do you want dialtone from the PRI. The NEC should send all the digits to Asterisk and asterisk will dial the call automatically. |
| 15:39.02 | WIMPy | And has been the major cause of complaints for me so far. |
| 15:39.19 | Assid | WIMPy: what has? no dialtone? |
| 15:39.45 | WIMPy | Ne, the timeout wehn using en-block. |
| 15:40.13 | igcewieling | Assid: you understand the PRI to your telco doesn't give you dialtone, right? |
| 15:40.36 | Assid | yes.. but if i directly connect the telco to the NEC.. it gets a dialtone SOMEHOW |
| 15:40.39 | Assid | when i hit 0 |
| 15:40.43 | WIMPy | Strange roumors in this channel. Off course the telcos PRI gives dialtone. |
| 15:41.26 | WIMPy | And offc ourese you could get your telcos dialtone through Asterisk. |
| 15:41.55 | Assid | wait.. how ??? |
| 15:42.47 | WIMPy | By using ! patterns and a normal Dial(). |
| 15:43.51 | Assid | can you show me an example please ? |
| 15:44.52 | WIMPy | If you use 0 for external, exten => _0!,1,Dial(dahdi/g<EXT>/${EXTEN}) |
| 15:46.13 | leifmadsen | that would also match dialing 0 |
| 15:46.16 | leifmadsen | is that what you would want? |
| 15:46.50 | Assid | so for this.. i need overlap dialling again ? |
| 15:47.03 | WIMPy | Sure. No overlap, no dialtone. |
| 15:47.54 | Assid | so in dahdi.. i need immediate=yes and overlapdial=yes on the nec port and the telco ? |
| 15:48.09 | igcewieling | I totally and utterly disagree with how WIMPy is recommending setting this up. |
| 15:48.12 | WIMPy | yes |
| 15:48.44 | igcewieling | leifmadsen: he wants Asterisk to provide dialtone to calls coming in on the PRI interface on Asterisk |
| 15:49.06 | WIMPy | That's probably because you only call national numbers in a close number plan country. |
| 15:49.27 | Assid | we do ALOT of international dialling.. |
| 15:49.33 | WIMPy | Some people want to call other countries as well. |
| 15:49.52 | WIMPy | And not be bothered with timeouts. |
| 15:50.03 | Assid | one of the guys has to call on an average 10 countries a day |
| 15:50.24 | igcewieling | Yes, overlap dialing for international does make some sense. Between two PBXs, not so much. However, it is the immediately=yes which I totally disagree with |
| 15:51.33 | WIMPy | Without immediate you just hide some part of the functionality from your dialplan. Where's the advantage of not knowing what's going on. |
| 15:51.33 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 15:51.52 | WIMPy | ? |
| 15:52.28 | Assid | WIMPy: so exten => s!,n,Dial(DAHDI/i1/${ARG1}) |
| 15:52.45 | igcewieling | Assid: no. |
| 15:52.48 | Assid | the outgoing uses a macro.. |
| 15:53.02 | WIMPy | No. S is just s and needs a WaitExten(). |
| 15:53.10 | Assid | exten => _9XXXXXXXXX,1,Macro(outgoing-airtel,${EXTEN}) |
| 15:53.21 | WIMPy | You can;t do it with a macro. |
| 15:53.27 | Assid | oh |
| 15:53.36 | igcewieling | Assid: when you use immediate=yes the ONLY extension which works is "s" |
| 15:54.02 | WIMPy | No, you don't have to hit s. |
| 15:54.13 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 15:55.16 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 15:55.26 | WIMPy | But many PBXs convert en-block to overlap, which is pretty stoopid, but then you will always go through s. |
| 15:55.43 | Assid | WIMPy: but even then we're unsure of it working |
| 15:55.47 | Assid | hmm |
| 15:56.14 | WIMPy | What do you mean by that? |
| 15:56.33 | Assid | using the ! dial method |
| 15:57.24 | Assid | when i enable overlapdial & immediate .. my nec was giving alot of wrong calls.. like i call this extension.. it ends up elsewhere.. |
| 15:57.33 | WIMPy | is not sure what the question is. |
| 15:57.35 | Assid | thats why i had to disable it.. and im using without that |
| 15:58.21 | WIMPy | Do you have any switches in your dialplan? |
| 15:58.53 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 15:59.01 | Assid | yes.. people who have an actual DID number show up as their full number.. everyone else gets the board line DID |
| 16:00.43 | WIMPy | Where does that fit in to the story? |
| 16:01.06 | Assid | if/switch ? |
| 16:01.07 | WIMPy | And does it imply some question? |
| 16:01.38 | WIMPy | The question was if you have a switch => statement in your dislplan. |
| 16:02.17 | Assid | nope |
| 16:02.20 | *** part/#asterisk leedm777 (~leedm777@nat/digium/x-kafaadvbfowokmdq) |
| 16:03.16 | WIMPy | Then something must go wrong at a lower level. |
| 16:03.47 | WIMPy | Show us a failed call. |
| 16:04.37 | Assid | asterisk side.. perfect.. its the NEC side that acts wonky |
| 16:05.47 | Assid | you know.. let me try and convince these guys this is the new dialtone |
| 16:05.55 | Assid | my problem would be solved immediately |
| 16:06.10 | jmetro | sounds like the solution |
| 16:06.30 | Assid | its just a busy tone tho :D |
| 16:08.22 | *** join/#asterisk madhatt (~madhatt@23.31.65.29) |
| 16:08.24 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-kafaadvbfowokmdq) |
| 16:08.26 | madhatt | hey all |
| 16:09.10 | WIMPy | Busy? |
| 16:09.17 | madhatt | got a questions… I'm going testing with SIPVicsious and I see that if I hit one of my Asterisk PBXs it responds and SVmap finds it. but the other doesn' and I can't figure out why? Of course I want both of my servers to NOT respond to svmap. any help out there? |
| 16:09.18 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 16:11.49 | Assid | WIMPy: if i use enbloc .. theres no waitexten is there ? |
| 16:11.59 | WIMPy | no |
| 16:12.49 | Assid | i wonder if the NEC has a digit timeout from last button dialled.. its taking a while.. and i dont want issues with international calling |
| 16:14.18 | WIMPy | If you tell the NEC to do enblock, it has to do a timeout. Otherwise Asterisk will do it. |
| 16:14.34 | igcewieling | sounds like your NEC should be providing dialtone |
| 16:15.06 | Assid | it gives me an initial dialtone.. |
| 16:15.41 | igcewieling | why do you need more than one dialtone. let the NEC collect the digits |
| 16:16.20 | polysics | I might be doing something wrong here |
| 16:16.29 | WIMPy | The timeout is the trouble. |
| 16:16.42 | Assid | yeah.. i saw this guy dialling 26 digits once |
| 16:16.52 | polysics | how do I make it so that a person in a conference hears "someone joined" chime while another does not? |
| 16:17.16 | polysics | I tried using two diff profiles with quiet=yes nd no but it seems to simply quiet the whole conf |
| 16:17.27 | WIMPy | Users do both complain about the waiting time as well as being aborted when dialling too slowly. It's a real PITA. |
| 16:17.54 | Assid | yeah i can see it happening already |
| 16:19.28 | Kobaz | so |
| 16:19.39 | Kobaz | what's a good way to do a string replace in dialplan in 1.8 |
| 16:19.40 | Assid | any idea what field i should be looking for on the NEC for the timeout on this? |
| 16:19.52 | Kobaz | 11 has the handy dandy STRREPLACE |
| 16:20.05 | igcewieling | Kobaz: give us sample string |
| 16:20.08 | Kobaz | i have a feeling i need to do something silly with CUT and ARRAY |
| 16:20.18 | Kobaz | just escape out commas |
| 16:20.24 | *** join/#asterisk tangledhelix (~dan@99-59-105-172.lightspeed.bcvloh.sbcglobal.net) |
| 16:20.25 | *** join/#asterisk navaismo (~navaismo@189.241.34.22) |
| 16:20.35 | Kobaz | foo,bar becomes foo\,bar |
| 16:20.59 | Kobaz | or. i can backport strreplace to 1.8 |
| 16:21.01 | Kobaz | that might be better |
| 16:21.07 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 16:21.08 | Kobaz | rather than wasting time doing it very manually |
| 16:22.16 | igcewieling | I wonder if any of the QUOTE* functions will escape commas |
| 16:22.33 | Kobaz | actually i don't think i can do it in straight dialplan |
| 16:22.47 | Kobaz | there's no character counting functions that i see |
| 16:23.45 | Kobaz | backport it is |
| 16:24.11 | igcewieling | FIELDNUM ? |
| 16:24.23 | igcewieling | Kobaz: wow, you like pain and suffering |
| 16:24.35 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 16:24.39 | Kobaz | why |
| 16:24.47 | WIMPy | Assid: T302? |
| 16:25.02 | Kobaz | wouldn't calling STRREPLACE be way easier than writing some character by character string processer in dialplan? |
| 16:25.28 | Assid | wait i saw that somewhere |
| 16:25.31 | igcewieling | yes, but keeping your patches current with new versions of asterisk is a lot of work. |
| 16:25.39 | Kobaz | FIELDNUM wont help you count how many things there are in a string |
| 16:25.46 | Kobaz | igcewieling: not really |
| 16:25.55 | Kobaz | igcewieling: i ported all my 1.8 stuff to 11 in 4 days |
| 16:26.04 | Kobaz | most of it applied cleanly |
| 16:26.11 | igcewieling | for features which exist in later versions of Asterisk I usually write a small AGI |
| 16:26.17 | Kobaz | i had 20 small sections of code to manually merge |
| 16:26.34 | Kobaz | and some random one liners here and there as well |
| 16:28.49 | Kobaz | bing |
| 16:28.56 | Kobaz | backported in 3.5 minutes |
| 16:29.20 | [TK]D-Fender | [12:22]Kobazthere's no character counting functions that i see <- this is a simple expression. |
| 16:30.35 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 16:31.05 | Kobaz | how simple? |
| 16:31.17 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 16:31.30 | Assid | WIMPy: nah.. ddidnt follow that timer |
| 16:32.18 | [TK]D-Fender | Kobaz: length of str - length of STR_REPLACE'd stripping of char to count |
| 16:32.45 | Kobaz | oh, yeah that works, heh |
| 16:32.50 | WIMPy | Assid: That was my best bet. I don't think there's an official time for the exact purpose. |
| 16:32.50 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 16:33.05 | Kobaz | still, backporting STRREPLACE was a lot faster than writing a STRREPLACE in dialplan |
| 16:33.24 | Assid | hehe.. k |
| 16:34.45 | polysics | sorry for the repost, but is it correct to assume that if a confBridge user has quiet=yes in his profile and another quiet=no, only the second hears the "person joined" chime? |
| 16:35.01 | polysics | or am I flipping things here? |
| 16:35.07 | Kobaz | [TK]D-Fender: you would need to get the number of items, start building a new string by appending on results from CUT along with your replacement character... ick |
| 16:35.23 | *** join/#asterisk jonniemac315 (~Jonalanni@99-32-51-201.uvs.toldoh.sbcglobal.net) |
| 16:35.57 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 16:36.02 | Assid | according to google i should be looking at T302 |
| 16:37.01 | [TK]D-Fender | Kobaz: What is it you are looking to do exactly? |
| 16:37.09 | Kobaz | escape out commas |
| 16:37.25 | [TK]D-Fender | Kobaz: as in just add \? |
| 16:37.32 | Kobaz | \, |
| 16:37.33 | Kobaz | yeap |
| 16:37.39 | Kobaz | there's no STRREPLACE In 1.8 |
| 16:37.42 | [TK]D-Fender | Kobaz: the a single STRREPLACE does that |
| 16:37.58 | Kobaz | and i don't feel particularly included to write a strreplace in dialplan |
| 16:38.07 | Kobaz | inclined... rather |
| 16:38.22 | [TK]D-Fender | Kobaz: "core show function REPLACE" <- |
| 16:38.28 | [TK]D-Fender | 1.8 |
| 16:38.28 | Kobaz | that only does one character |
| 16:38.35 | Kobaz | which is why you need strreplace |
| 16:38.50 | WIMPy | There's no life in 1.8, either. |
| 16:38.54 | [TK]D-Fender | Kobaz: the instructions say SET of chars |
| 16:39.07 | igcewieling | There is plenty of life in 1.8 it is still in LTS |
| 16:39.08 | Kobaz | Replace a set of characters in a given string with another character. |
| 16:39.10 | Kobaz | yes |
| 16:39.16 | Kobaz | replace a "set of" characters |
| 16:39.24 | Kobaz | with "another" character meaning one |
| 16:39.27 | [TK]D-Fender | Kobaz: Have you tried it? |
| 16:39.35 | WIMPy | igcewieling: You call that a life? |
| 16:39.45 | [TK]D-Fender | Kobaz: and I do see the MATCH is many, the replace is ONE... which I am doubting. |
| 16:39.48 | igcewieling | WIMPy: yes. |
| 16:39.56 | madhatt | anyone here familiar with sipvicious? |
| 16:39.58 | Kobaz | it's replace one |
| 16:40.06 | [TK]D-Fender | Kobaz: you've tested? |
| 16:40.18 | igcewieling | madhatt: Is that aka Friendly-Scanner? |
| 16:40.27 | Kobaz | [TK]D-Fender: i've looked at the code |
| 16:40.37 | madhatt | yup, I'm trying to understand why one of my pbx responds to it the both other doesn't.... |
| 16:40.41 | Kobaz | <PROTECTED> |
| 16:40.42 | Kobaz | <PROTECTED> |
| 16:41.17 | Kobaz | just enough storage for one character and a trailing null |
| 16:41.49 | WIMPy | What about using AGI? |
| 16:41.53 | igcewieling | WIMPy: as long as the branch is getting updates it has life |
| 16:42.12 | Kobaz | WIMPy: i just backported STRREPLACE to 1.8, so i'm good |
| 16:42.32 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 16:48.02 | Assid | hmm.. i should probably check out getting video working through asterisk |
| 16:49.05 | Assid | alrite.. thanks guys.. ciao |
| 16:53.20 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:58.54 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 17:03.38 | Kobaz | okay |
| 17:03.39 | Kobaz | i dood it |
| 17:03.42 | Kobaz | <PROTECTED> |
| 17:03.58 | Kobaz | it would be nice if dialplan had actual parameter placeholders |
| 17:04.10 | Kobaz | and it wasn't just one long string |
| 17:04.36 | jmetro | hu |
| 17:04.37 | jmetro | h |
| 17:06.11 | Kobaz | escape out the commas |
| 17:06.44 | Kobaz | because if you do UserEvent(Foo,MyCallerid:${QueueCalleridName},Otherdata:bar) without escaping |
| 17:07.50 | Kobaz | say your callerid is "jones, bob"... you can wind up with [UserEvent: Foo] [MyCallerid: jones] [bob: otherdata] [bar] |
| 17:08.22 | Kobaz | i didn't realize i had this bug for ages |
| 17:08.33 | Kobaz | it's probably been unescaped like that for a year |
| 17:09.30 | Kobaz | if dialplan had more standard parameter handling you wouldn't wind up with problems like that |
| 17:09.54 | Kobaz | commas inside your strings shouldn't affect parameter placement |
| 17:18.04 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 17:21.11 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 17:30.49 | *** join/#asterisk emamdouh (~emamdouh@196.219.211.1) |
| 17:31.35 | emamdouh | hi i'am newbie to asterisk , i've plugged in two land lines to a sangoma analog card |
| 17:32.00 | emamdouh | I was trying to execute command dahdi_scan 2 3 4 |
| 17:32.19 | emamdouh | but with no outpu |
| 17:32.23 | emamdouh | output* |
| 17:32.37 | emamdouh | however dahdi_scan 1 result in some output |
| 17:32.54 | navaismo | Didi you installed the sangoma software(wanpipe)? Did you ask support for the vendor? |
| 17:33.33 | emamdouh | actually this systems was running before with 4 land lines |
| 17:33.42 | emamdouh | we just moved server to another location |
| 17:33.52 | emamdouh | and we've attached another two land lines |
| 17:34.16 | emamdouh | my problem is that just one of them is working, however second one isn't |
| 17:34.41 | emamdouh | that's why i'm asking , is that a sign that 2nd line has a physical issue ? |
| 17:34.52 | emamdouh | so i need to check it physically |
| 17:39.31 | *** join/#asterisk angryuser_laptop (~angryuser@ip-189.net-81-220-67.lyon.rev.numericable.fr) |
| 17:40.23 | *** join/#asterisk navaismo (~navaismo@189.241.34.22) |
| 17:41.42 | navaismo | is configured the second line? |
| 17:42.15 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 17:42.23 | asilva | navaismo: hey there, i'm making tests between same versions, i'll have to open a bug on jira, couldn't figure it out what might be! |
| 17:42.47 | *** join/#asterisk envirocbr (~enviro@unaffiliated/envirocbr) |
| 17:43.03 | envirocbr | Well, TWCBC sucks balls, they can only deliver my a 23 channel PRI for a massive sum of money :/ |
| 17:43.15 | envirocbr | really wants to just host this |
| 17:43.25 | envirocbr | Any reliable VoIP hosts? |
| 17:44.27 | *** join/#asterisk dms (~dms@65.207.151.254) |
| 17:44.33 | dms | Hi Folks! |
| 17:44.41 | dms | been a while since I've asked a question here |
| 17:44.44 | dms | lots of years |
| 17:44.45 | dms | ;) |
| 17:45.13 | dms | anyone know if a TCE400 supports pcie passthru in vmware ? |
| 17:46.33 | dms | i.e. Intel VT-d / DirectPath I/O |
| 17:46.43 | navaismo | asilva, again: weird<goat voice> |
| 17:47.03 | asilva | ehehehe i know, just informing ehhehe!! collecting debugs now |
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| 17:52.41 | jmls | afternoon |
| 17:53.11 | jmls | if I am using local/foo@context as my "device" for queues, would that cause problems with the wrapup time ? |
| 17:53.23 | jmls | it doesn't seem to be honoured |
| 17:53.57 | jmls | I am thinking that because it is a local channel that is "deleted" after connecting to a real sip device |
| 17:54.19 | jmls | [optimised - that's the word I was thinking about |
| 17:54.26 | igcewieling | try adding /n to context |
| 17:54.41 | jmls | that's what I was thinking. just wanted to confirm |
| 17:55.10 | jmls | I just get to have another channel active for the call, and didn't want to have that overhead |
| 17:55.22 | jmls | but if there's no other option, then .. |
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| 18:06.01 | leifmadsen | jmls: what does the whole line look like? Are you using SIP peers as the underlying tech? If so, are you giving the queue the device to monitor state from? |
| 18:06.33 | leifmadsen | jmls: e.g. ;member => Local/1000@default,0,John Smith,SIP/1000 |
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| 18:34.13 | gnudna | is there a reason why cisco 7960 do not work with asterisk 1.8? |
| 18:34.30 | gnudna | would not work is what i meant |
| 18:34.41 | gnudna | im having a hard time getting thm to register |
| 18:35.14 | file | they do work, but their SIP stack can be very finicky and the behavior changes based on the firmware version |
| 18:35.18 | [TK]D-Fender | gnudna: Then you set it up wrong |
| 18:35.22 | file | you also can't use nat=yes with it, it must be set to nat=no |
| 18:37.00 | drmessano | and you pay Cisco for that privledge |
| 18:37.49 | drmessano | "What do I get with a TAC?" "Lighter wallet, bragging rights. You could also buy a brick from Lowe's, which would have the same effect" |
| 18:41.02 | igcewieling | If you have other cisco equipment (real Cisco not Linksys) having IOS updates can be handy |
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| 18:43.02 | drmessano | For routers, switches, and AP's, sure. Phones. Um yeah |
| 18:46.59 | gnudna | [TK]D-Fender, they working on asterisk 1.6 |
| 18:47.07 | gnudna | so what changed in asterisk 1.8? |
| 18:47.23 | jmetro | wow ast 1.6? i will have to upgrade im on 1.2 |
| 18:47.29 | [TK]D-Fender | gnudna: Nothing |
| 18:47.43 | gnudna | so how can i have them configured wrong i did not touch them |
| 18:47.54 | gnudna | the polycoms work without issue |
| 18:48.38 | igcewieling | gnudna: did you check the UPGRADE-*.txt files for anything which might be related. I seem to recall a bug with Asterisk involving Asterisk which was fixed, but I don't remember where. What EXACT version if Asterisk are you using now? |
| 18:50.27 | gnudna | asterisk 1.8.13.1~dfsg-1~bpo60+1 |
| 18:50.34 | gtTuna | what do people generally use for call quality monitoring? |
| 18:50.45 | gtTuna | preferably something somewhat automated |
| 18:52.58 | igcewieling | gnudna: stop wasting your time and update to the latest Asterisk |
| 18:53.04 | igcewieling | (latest Asterisk 1.8) |
| 18:54.11 | jmetro | call quality monitoring? Every human comes with two standard equipped. |
| 18:54.50 | igcewieling | our method is to wait for people to bitch about call quality, then go into the local Adtran media gateway and check the call quality stats |
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| 18:55.09 | slav3_kitten | jmetro, what about deaf people? |
| 18:55.35 | jmetro | slav3_kitten: Overlooked on the production line, send back for a full refund plus a complimentary tote. |
| 18:55.51 | gtTuna | i mean... |
| 18:55.58 | gtTuna | i can't listen to every call |
| 18:55.59 | slav3_kitten | wooooooooo i can send my deaf friends back. do i get like a better deal if i send 5 at a time? |
| 18:56.44 | igcewieling | slav3_kitten: did you keep the receipt? |
| 18:56.49 | slav3_kitten | pretty sure |
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| 19:00.28 | drmessano | You don't need the receipt at Wal-Mart.. they will give you store credit, loaded onto a lead-based plastic card, so you can buy more lead-based products |
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| 19:07.24 | jmetro | drmessano: i love lead, it tastes great on chips. |
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| 19:30.22 | coppice | we don't use lead to solder our chips any more |
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| 19:35.22 | polysics | is there a way to know where an Asterisk instance gets its peers? |
| 19:35.33 | polysics | sip.conf, realtime, res_ldap, whatever? |
| 19:35.37 | Kobaz | /etc/asterisk |
| 19:35.42 | Kobaz | depends on how you have it set up |
| 19:35.52 | polysics | yeah, it's difficult to tell here |
| 19:36.01 | pabelanger | *CLI> module reload chan_sip, see what gets parsed |
| 19:36.32 | Kobaz | /etc/asterisk/extconfig.conf |
| 19:51.49 | polysics | Kobaz: file is completely commented so I can assume it's sip.conf. Thanks! |
| 19:52.13 | Kobaz | another easy easy way to check |
| 19:52.25 | Kobaz | is add something to sip.conf, sip reload... see if it shows up |
| 19:58.24 | polysics | does adding call-limit=1 need an Asterisk restart? or just sip reload? |
| 19:58.49 | igcewieling | polysics: almost nothing requires an Asterisk restart. |
| 19:59.02 | polysics | MoH changes seem to |
| 19:59.26 | igcewieling | module unload res_musiconhold.so may work as well |
| 20:03.25 | jmetro | hm, what does :wqa do |
| 20:03.30 | jmetro | write quit... ? |
| 20:03.37 | jmetro | im stupid and should use google. |
| 20:07.43 | igcewieling | -1 @jmetro |
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| 21:46.28 | polysics | is it possible that a admin exiting the conference and coming back in loses the admin menu? |
| 21:46.52 | polysics | ConfBridge, admin hangs up by mistake, redials, gets dropped in but no menu |
| 21:50.07 | mjordan | polysics: only if you don't give the participant the menu again |
| 21:50.34 | mjordan | polysics: if they hang up and dial back in, they had to go back in through the dialplan. That means they had to execute the ConfBridge application again, which is where the DTMF menu is applied |
| 21:51.06 | polysics | yes, nvm, I think we have a logic failure here |
| 21:51.32 | polysics | they are coming back in with the wrong profile |
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| 21:58.05 | polysics | mjordan: no, logic is correct. We are using admin_kick_last to kick the customer. Is it possible that the customer is not the last in the conference, since the agent re-logged back in? |
| 21:59.41 | mjordan | admin_kick_last kicks the last participant in the conference. It doesn't pick and choose. |
| 22:02.03 | polysics | so if I exit and go back in, I become the last in the conference, = it stops working? |
| 22:03.48 | mjordan | it doesn't stop working, it's doing exactly what you told it to - try to kick the last participant. And if you're the last participant (or if the last participant is also an admin, actually) it won't kick you. It will play back the error sound if you've defined it. |
| 22:05.47 | mjordan | if you want to kick a specific user, you can create that functionality by using the dialplan_exec menu option. You could bounce out to the dialplan, then use the CONFBRIDGE_INFO function to look up the user to eject, and ChannelRedirect them to wherever you want them to go |
| 22:06.55 | polysics | yeah, sorry, I mis-worded that. Is there a way to get that key to always kick the non-admin? |
| 22:07.07 | mjordan | no |
| 22:07.18 | mjordan | it always looks to the last participant, and attempts to kick them |
| 22:07.25 | mjordan | it doesn't search the list for the last non-admin. |
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| 22:09.59 | polysics | I can do it in Adhearsion logic |
| 22:10.13 | polysics | great, thanks a bunch. You saved me a lot of time |
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| 22:28.54 | polysics | there is no "kick everyone" function either, it seems |
| 22:29.05 | polysics | ah well, time for some good old dialplan logic :-) |
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| 22:37.36 | gusto | hey, i have a practical question |
| 22:37.41 | gusto | WIMPy: hi, are you there? |
| 22:38.09 | gusto | is there a way to tell asterisk that he should take a round robin random peer from a list or a context to make a call? |
| 22:38.53 | gusto | like when a VoIP provider has more VoIP-to-PSTN gateways and i do not want to use everytime the same IP |
| 22:39.23 | gusto | maybe they should think about some implementation of anycast, i know, but i am not responsible for their failures |
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