| 00:02.58 | Greenlight | You played around with the JITTERBUFFER dialplan command in Asteriks 10 much? |
| 00:16.24 | p3nguin | I've never touched Asterisk 10. |
| 00:16.42 | p3nguin | I didn't even migrate to 1.8 until a few months ago. |
| 00:26.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:27.39 | dijib | did you fix your sync issue eyet? |
| 00:52.05 | *** join/#asterisk datarecal (~datarecal@S0106c43dc7876e60.cg.shawcable.net) |
| 00:52.31 | datarecal | hello |
| 00:53.09 | datarecal | I keep getting this error today : chan_sip.c:12654 sip_reg_timeout: -- Registration for '1962921343@did.voip.les.net' timed out, trying again (Attempt #303). Dont know why , any thoughts ? |
| 00:54.22 | datarecal | cant find anything in google about this error |
| 00:56.08 | [TK]D-Fender | datarecal: Means the other side isn't responding. Either your networking is bad, your sip.conf is configured wrong, or they are down. |
| 00:56.19 | [TK]D-Fender | datarecal: pastebin the SIP debug of your attempts |
| 00:56.22 | [TK]D-Fender | ~pb |
| 00:56.23 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 00:58.36 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:58.42 | datarecal | http://paste2.org/p/1795376 |
| 01:03.03 | [TK]D-Fender | datarecal: PB "iptables --list" |
| 01:05.26 | datarecal | http://paste2.org/p/1795384 |
| 01:07.46 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 01:08.36 | [TK]D-Fender | datarecal: ok, check that 24.64.121.78 is still your proper WAN IP for NAT settings, ensure that the proper port ranges are forwarded. |
| 01:08.46 | [TK]D-Fender | datarecal: Other than that things look OK at the server level |
| 01:11.08 | [TK]D-Fender | Out for a while... |
| 01:20.48 | datarecal | [TK]D-Fender what ports are supposed to be forwarded ? |
| 01:20.53 | datarecal | .78 ip is my public ip |
| 01:21.56 | datarecal | http://screencast.com/t/7tjQ56aQqE are the router settings |
| 01:24.16 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-hyyhozgrxgbirdfz) |
| 01:32.43 | *** join/#asterisk adolfomaltez (~taro@190.62.214.63) |
| 01:51.35 | *** join/#asterisk e7e5 (~rudenko@188.134.34.92) |
| 02:05.13 | *** join/#asterisk scubes13 (~scubes13@66-191-209-216.dhcp.spbg.sc.charter.com) |
| 02:08.21 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-ufkcvwtynyaqapwh) |
| 02:20.56 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-lvcgckpwhbsoqxrx) |
| 02:21.25 | Kobaz | teh turkeys |
| 02:25.37 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279682108.dsl.bell.ca) |
| 02:25.53 | dijib | http://pastebin.com/Fe7mUn53 can anyone help me with this? |
| 02:26.00 | dijib | back in ten min going for a smoke |
| 02:26.21 | dijib | errors are in the second half of the PB |
| 02:35.19 | Kobaz | digiv: your syntax is all wrong apparently |
| 02:35.33 | Kobaz | start putting echo statements around to see which lines are causing problems |
| 02:35.51 | Kobaz | i haven't played with tc in forever... try #networking |
| 02:38.23 | SeRi | dijib, waz up |
| 02:44.39 | dijib | SeRi, im going to commit harry kery here this shaping shit is pissing me off, something about a rate function i assume on a iptables command and my ip-ranges arnt complying |
| 02:44.48 | dijib | fuck punctuation |
| 02:44.58 | dijib | need water |
| 02:46.20 | SeRi | dijib, I warned you. :P |
| 02:46.29 | SeRi | you in your conf? I can probabley help |
| 02:49.24 | SeRi | dijib, ^^ |
| 02:50.05 | dijib | oh yes i am |
| 02:50.15 | dijib | i think i can fix this though |
| 02:50.37 | SeRi | dijib, give me 10 ill jump in and talk to you about it. |
| 02:50.51 | dijib | let me see if i can make any progress |
| 02:51.26 | dijib | tc command |
| 02:52.52 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 03:10.37 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
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| 03:35.24 | p3nguin | quacks like a duck |
| 03:37.27 | SeRi | waz up p3nguin |
| 03:38.13 | SeRi | http://wiki.openwrt.org/doc/uci/qos |
| 03:38.23 | p3nguin | I think I ate too much. |
| 03:38.50 | SeRi | p3nguin, you and me both d00d |
| 03:41.13 | SeRi | p3nguin, I am helping dijib with hes script issues.... |
| 03:46.50 | *** join/#asterisk master_of_master (~master_of@p57B521A9.dip.t-dialin.net) |
| 03:47.06 | SeRi | p3nguin, you avail? |
| 03:51.47 | *** join/#asterisk Sedorox (brandon@smartserv/cna/Sedorox) |
| 03:54.05 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 04:08.11 | p3nguin | Okay. |
| 04:12.36 | SeRi | p3nguin, can you give us a hand here? |
| 04:14.44 | p3nguin | Not unless you ask me a question about something specific. |
| 04:25.11 | *** join/#asterisk irroot (~gregory@197.104.89.161) |
| 04:28.47 | SeRi | p3nguin, Ill let dijib ask than. He is having some issues with a script that looks like is not built for hes env. |
| 04:31.32 | SeRi | I wont bother you :) |
| 04:43.57 | dijib | Seri you dropped |
| 04:44.00 | SeRi | p3nguin, how was TK? |
| 04:44.05 | SeRi | I did? |
| 04:44.07 | dijib | yes |
| 04:44.10 | dijib | i hear moh |
| 04:45.29 | SeRi | dijib, I hear bithing |
| 04:45.34 | SeRi | nothing* |
| 04:45.41 | dijib | nothing? |
| 04:45.59 | SeRi | I hear like somebody scrambled |
| 04:46.30 | ChannelZ | that's just pig latin |
| 04:46.31 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 04:54.36 | irroot | ok now im sticking it to the bank :P offered to show them how i can decode DTMF into human readable format from a audio stream and or net stream |
| 04:55.16 | irroot | there call centre is asterisk no TLS/SRTP and they request customer pin codes via DTMF as proof of identity |
| 04:55.37 | *** join/#asterisk radic (~radic@dslb-178-002-234-034.pools.arcor-ip.net) |
| 05:01.32 | SeRi | irroot, stick it to them! they sure like to stick it to there customers! :P |
| 05:02.29 | irroot | there CEO has joined twiter as a PR thing so called him out on there |
| 05:03.47 | irroot | SeRi i dont think anyone here will think DTMF is secure to transfer ID/PIN combo in the states it would be the social security num |
| 05:04.23 | florz | where call center and where CEO? |
| 05:05.22 | SeRi | irroot, I agree I just find it funny. :) Bad practice to begin with.... :/ |
| 05:08.08 | irroot | florz the call centre is in johannesburg but its all IP with regional nodes the CEO is 20km from where i am now at HO |
| 05:08.10 | irroot | when you call the call centre the IVR asks for ID number then a pin code from one of your accounts it validates it in background |
| 05:08.12 | irroot | the problem is the DTMF is readable in IP packets and if your calls are been recorded that recording will be able to be decoded by piping |
| 05:08.14 | irroot | the wav through a dsp processor |
| 05:08.40 | florz | and where customers? |
| 05:11.29 | irroot | florz country wide and possibly out of country |
| 05:11.52 | florz | hint: you are constantly misspelling "their" |
| 05:12.19 | irroot | cant spell :P |
| 05:13.18 | SeRi | I do the same shit "there" "their" "these" "this" |
| 05:32.30 | dijib | http://pastebin.com/iqK0ZBf4 |
| 05:36.32 | irroot | dijib tc is your friend :P |
| 05:36.41 | irroot | http://lartc.org |
| 05:36.44 | *** join/#asterisk ajkaanbal (~ajkaanbal@201.102.183.96) |
| 05:40.07 | SeRi | I really really really and really think he should be using native qos/shapping to openwrt instead of building a script from scratch :) |
| 05:41.00 | SeRi | well if he wants it to just work.... if he likes to thinker and learn than that would do just fine. |
| 05:42.08 | irroot | SeRi yeah indeed when i added tc onto the distro it took me weeks i reakon to nail it down |
| 05:42.32 | SeRi | I spent at least two hours helping him clean the code. |
| 05:42.51 | SeRi | now at least it runs but its not define correctly |
| 05:42.55 | irroot | having multiple routing tables marking and clasifying packets with iptables then calculating and applying tc .... |
| 05:43.13 | irroot | it can hurt your heas |
| 05:43.17 | irroot | headf |
| 05:43.49 | SeRi | Indeed.... |
| 05:46.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:47.07 | irroot | i was kind to the M$ retards did a java applet that would do it for you build stateful firewall with tc and routing to destination here data is priced per Gb and |
| 05:47.09 | irroot | "local" only is much cheaper so we route email down the cheaper link also we get shaped/unshaped so we can priorotise using the old "TOS" flags |
| 05:48.00 | SeRi | lol |
| 05:48.05 | irroot | now we use a additional link for private IPC / DSL on 10. address space for VOIP and access to Our hosted enviroment at ITSP |
| 05:48.29 | SeRi | nice |
| 05:48.33 | irroot | so most will use 3 DSL connections Local/International/Voice |
| 05:49.37 | SeRi | nice |
| 05:49.49 | irroot | then we double down when leased lines cost too much we VLAN and have one DSL line per VLAN and load balance calls over all voice links |
| 05:50.23 | SeRi | irroot, I wanted to do something similar at home with vlans |
| 05:50.32 | SeRi | the switches where a bit to expensive though |
| 05:50.32 | irroot | a leased line will cost 10k per 2M that is 1000E whereas DSL costs +/- 1k per line so 300E compared to 1kE |
| 05:50.42 | SeRi | damn! |
| 05:51.49 | irroot | so you can see why doing it this way is more than just nice it saves big money |
| 05:53.16 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 05:53.53 | SeRi | I see. d00d wow |
| 06:00.24 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 06:03.53 | irroot | SeRi you dont need full L3+POE get the basic netgear / Linksys [CISCO] / Dlink for cheap here +/- 1k5 thats 150E |
| 06:04.19 | irroot | there is a edimax 8port + POE that comes in at same price |
| 06:05.15 | SeRi | irroot, yes I been contemplating the thought. |
| 06:05.28 | SeRi | I looked at some gigabit zyxel with poe and vlan |
| 06:06.51 | irroot | yeah about the same i have not seen zyxel for a while must find out why .... |
| 06:09.22 | p3nguin | Off to the Black Friday sales I go! |
| 06:09.34 | SeRi | p3nguin, good luck |
| 06:09.37 | SeRi | have fun |
| 06:10.23 | irroot | p3nguin hehe i must write a thesis on this some time trying to explain how its a self fulfilling prophecy but if you got the loot |
| 06:10.25 | irroot | take advantage while you can |
| 06:11.40 | SeRi | lol |
| 06:13.14 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
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| 06:57.28 | p3nguin | I got a Shop Vac and a Monopoly board game. |
| 06:57.40 | SeRi | That fast? |
| 06:57.51 | p3nguin | Walmart isn't too far away. |
| 06:57.55 | SeRi | nice! |
| 06:58.17 | SeRi | shit there was lines at the walmart over by my house... |
| 06:58.43 | SeRi | there was people camping for like two days at the best buy parking lot |
| 06:59.37 | p3nguin | I probably didn't spend 20 minutes total at Walmart. I got in, got my stuff, and got out. |
| 06:59.55 | *** join/#asterisk LostyJai (~blah@202.171.190.130) |
| 06:59.56 | LostyJai | anyone here? |
| 07:00.05 | LostyJai | where do you put the trunk settings |
| 07:00.13 | p3nguin | iax.conf? |
| 07:00.19 | LostyJai | thanks again p3nguin |
| 07:00.42 | SeRi | p3nguin, nice. |
| 07:01.04 | p3nguin | I'll probably spend a little longer when I go to Radio Shack later. |
| 07:01.18 | p3nguin | But again, only two things on my list for that store. |
| 07:01.30 | SeRi | you going tonight? |
| 07:01.41 | p3nguin | They open at 0530. |
| 07:01.42 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:01.53 | SeRi | ah I see. |
| 07:02.37 | SeRi | My wanted to go shopping for xmas gifts and I said no.... |
| 07:02.52 | SeRi | to go pick up a few toys is not worth the damn hasle |
| 07:03.09 | p3nguin | You're doing it wrong! |
| 07:03.24 | p3nguin | Should have gone shopping over here at our Walmart. |
| 07:03.27 | LostyJai | hmmm |
| 07:03.35 | LostyJai | says ... [Nov 25 18:01:23] == Everyone is busy/congested at this time (1:0/0/1) |
| 07:03.37 | SeRi | last time I did it the line at kholes to pay wrapped around the store and the ladie in front of my shit it her pants |
| 07:03.37 | LostyJai | hmm |
| 07:03.46 | p3nguin | The toy section was pretty much empty of people. |
| 07:04.07 | SeRi | p3nguin, damn that must be nice. |
| 07:05.14 | p3nguin | Too bad they didn't sell air freshener, huh? |
| 07:05.21 | LostyJai | nvm... |
| 07:05.27 | SeRi | people do some stupid shit just to get some of this deals. when that ladie shit it her pants I said fuck this... drop the toys and left |
| 07:06.27 | SeRi | like 3 people behind me did the same thing.... put their shit down and left |
| 07:07.09 | SeRi | p3nguin, LOL I dont think air freshener would of have help. lol |
| 07:07.32 | SeRi | p3nguin, In what type of env would you use SER? |
| 07:08.06 | p3nguin | I'd probably use it in one where I have a ton of devices registering and where I need several asterisk systems to handle all the calls. |
| 07:08.16 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 07:08.52 | SeRi | p3nguin, I see. I all ways wanted to see what is SER all about but seems to be more for large deployments. |
| 07:09.06 | p3nguin | Look at milkfish. |
| 07:09.59 | SeRi | p3nguin, I came across them but I thought it was a dead end???? |
| 07:10.37 | p3nguin | I think milkfish is like a scaled down SER. I could be mistaken, since I have not used it, though. |
| 07:10.38 | SeRi | looks like they made a build for dd-wrt |
| 07:11.24 | p3nguin | There's also kamailio. |
| 07:12.34 | SeRi | I see. is SSIP or secure sip worth the hassle? |
| 07:13.07 | p3nguin | I suppose if you are worried about other people being able to see your traffic, maybe. |
| 07:13.48 | SeRi | got cha.... wow kamailio seems enterprise like.... |
| 07:13.49 | p3nguin | I personally don't care all that much. If some network nimrod wants to steal my SIP packets and do something stupid, so be it. |
| 07:14.39 | *** join/#asterisk qakhan (~qakhan@182.185.182.90) |
| 07:14.40 | SeRi | I hear you..... I doubt is actually done much unless theya re really after you.... in another word you must be a specific target |
| 07:14.45 | qakhan | hi all |
| 07:15.03 | qakhan | can someone tell about espeak |
| 07:15.11 | *** join/#asterisk irroot (~gregory@196-215-57-105.dynamic.isadsl.co.za) |
| 07:15.12 | p3nguin | I think I'm going to go have a nap for a while before I go to the other store. |
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| 07:16.34 | awk | hmm, is there a way to reset the local config of a polycom remotely, aka via links or w3m ? |
| 07:16.37 | SeRi | p3nguin, ok. g/n |
| 07:16.39 | awk | or via some 'notify' command |
| 07:17.28 | qakhan | <PROTECTED> |
| 07:21.10 | irroot | awk notify yes |
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| 07:23.24 | qakhan | irroot can u tell me abour espeak |
| 07:24.02 | kaldemar | qakhan: i doubt anyone is willing to lecture you about espeak in general. if you have a specific question, just ask it. |
| 07:26.45 | awk | yes, what is so difficult to understand about speak? |
| 07:26.51 | awk | err espeak ? |
| 07:29.46 | qakhan | yes i am installing espeak on CentOS 5.5 with asterisk 1.4.38 |
| 07:30.00 | qakhan | but i am fail to install it |
| 07:30.37 | qakhan | is ask to install dependencies which i tried but it says no espeak-devel available |
| 07:31.32 | qakhan | can u tell me what exact dependencies required for espeak with asterisk 1.4.38 on CentOS 5.5 |
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| 07:35.17 | GG111 | Greetings all, got a question about jabber.conf , in specific, I cannot get Asterisk to connect successfully to Google's XMPP |
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| 07:36.01 | jenna | join #ltsp |
| 07:36.34 | eicto | jenna, advestering ? :) |
| 07:36.41 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:36.46 | schmidts | good morning |
| 07:36.53 | jenna | eicto, hehe sorry |
| 07:37.05 | eicto | Good morning |
| 07:52.11 | wdoekes2 | morning all |
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| 07:57.40 | kaldemar | qakhan: ask centos people how to install espeak-devel |
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| 09:39.25 | IsUp | hi |
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| 10:01.50 | qakhan | plz help me to install sphinx wiht asterisk on centos 5.7 |
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| 10:06.30 | FlashDeluxe | hi! i am using asterisk 1.8 and dahdi 2.4.0 and in almost all my calls i got short dropouts. Does anybody can tell me how i can get rid of it e.g. which paramteres i do have to change? Or does anybody can tell me how to debug this? Thanks in advance! |
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| 10:21.35 | nunne | Anyone have any experience with mailing via "sendmail" and using MIME? (Not the correct channel, i know :P). But if i just send the attachtment after using uuencode and no MIME it works (but some mail clients doesnt like this)... But trying to use MIME it will "work" as far as getting a mail with an attachment, but the attachment is at 0kb in size |
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| 10:24.10 | wdoekes2 | nunne: sendmail is a very oold application. you might mean the /usr/lib/sendmail wrapper. as for attachments.. getting mime to work properly is non-trivial. but the are lots of libs that can do this for you.. and usually they will talk with the sendmail wrapper and/or the smtpd on 25 too |
| 10:26.11 | nunne | wdoekes2: im on an embedded platform using ssmtp. so my options are quite limited :) |
| 10:26.20 | wdoekes2 | and a hint: looking at the raw mail output of a mail with attachment generated by your desktop mail client, will give you necessary clues |
| 10:26.23 | nunne | http://pastebin.com/AXeBG0hV <- this works, with my mailapp without MIME |
| 10:26.40 | nunne | http://pastebin.com/JvYr5yR3 <- does not work.. and neither does this: http://pastebin.com/rWdfPMU5 |
| 10:26.49 | wdoekes2 | yuck.. |
| 10:27.03 | wdoekes2 | can you apt-get the biabam shell script? |
| 10:27.18 | wdoekes2 | it's the bash attachment mailer |
| 10:28.04 | nunne | i dont even have bash :) I just have hush. so not all shell scripts works for me :/ |
| 10:28.08 | wdoekes2 | in JvY: you're missing mime headers at the top |
| 10:29.16 | wdoekes2 | eeew.. and you're setting content type to base64 while putting uuencoded stuff with filename garbage in it |
| 10:29.24 | wdoekes2 | how can you expect that to work? |
| 10:30.01 | wdoekes2 | that's simply gross |
| 10:30.09 | wdoekes2 | biabam is only 160 lines |
| 10:30.19 | wdoekes2 | if it isn't hush compatible, you can at least look at how it works |
| 10:33.33 | nunne | haha, i suck at everything email related basically ;) |
| 10:33.59 | nunne | biabam was not hush compatible since it wanted to store things in an array. but i will have a look at how it works |
| 10:35.00 | qakhan | i am new in asterisk and sphinx. please help me how install sphinx with asterisk and setup |
| 10:35.42 | nunne | Ahaaa, yeah. it has all the info easily avaible in the script :) thanks wdoekes2 |
| 10:36.00 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 10:37.27 | IsUp | nunne: maybe you can use 'mutt' |
| 10:47.45 | nunne | IsUp: the system doesnt have mutt, so im stuck with making a shell script. its a super simple case of just mailing an attached file. doesnt have to be fancy at all really. |
| 10:48.23 | IsUp | nunne: good luck :p |
| 10:48.34 | nunne | thanks :) |
| 11:01.28 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 11:04.12 | nunne | i found whats wrong.. uuencode doesnt want to do base64 enconding correctly for me :( |
| 11:12.19 | wdoekes2 | uuencode -m |
| 11:12.21 | wdoekes2 | nunne: |
| 11:13.44 | wdoekes2 | but there's a base64 in debian coreutils, you can use that one instead |
| 11:24.15 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 11:29.58 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
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| 11:35.30 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 11:46.12 | ixyd_ | hi! |
| 11:46.49 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 11:46.56 | IsUp | ixyd_: hello |
| 11:48.55 | ixyd_ | while debugging a sip-refer problem in combination with an sbc, i found that asterisk is sending "SIP/2.0 183 Ringing" in a sipfrag-body in corresponding SIP-Notifies...the sbc responds to this by "400 Bad Request" - i think this is a bug and should be 180 Ringing instead...any comments on this? :) |
| 11:51.34 | *** join/#asterisk gnude (~gnude@p578bb787.dip0.t-ipconnect.de) |
| 11:54.45 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:11.33 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 12:11.37 | jacc0 | hi all!! |
| 12:11.48 | jacc0 | it's almost weekend over here!! |
| 12:11.58 | jacc0 | 5 hours to go |
| 12:12.02 | jacc0 | :) |
| 12:12.20 | IsUp | jacc0: same here. |
| 12:18.35 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
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| 12:39.25 | *** join/#asterisk Diffen (~diffen@c-a27ce555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 12:42.28 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:45.52 | Diffen | Hello all. Im trying to connect two asterisk togheter via SIP. My plan is that server 1 that are connected to the PSTN network should be a sip provider to my test server. My problem is that i cant register server 2s trunk on server 1. It says cant find matching peer on server 1. here are my confs. http://pastebin.com/JLwUtSmT Server 1 is version 1.4 and Server 2 1.8 |
| 12:51.10 | IsUp | Diffen: what says 'sip show registry' ? |
| 12:52.52 | IsUp | Diffen: and yes, theres no matching peer on server 1, set [Server1] to [test] |
| 12:52.55 | Diffen | on server 2 it says rejected |
| 12:52.58 | kaldemar | Diffen: test:test should be Server1:test |
| 12:53.05 | Diffen | ok |
| 12:53.29 | kaldemar | or Server2:test, you didn't say which configs are on which server and was that all your config and was it real. |
| 12:54.09 | IsUp | Diffen: also make sure your register => line is under [general] |
| 12:54.32 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 12:57.00 | tuxx- | hi guys. Is asterisk able to dial a sip string? What im looking for specifically is that i want to dial to different sip domains. Currently i have a sip trunk setup between opensips and asterisk, but that only tries to dial to the domain in the 'host' variable of the siptrunk. I would like to dial to: sip:<exten>@<domain>. |
| 12:57.06 | Diffen | ok so my register line on server 2 should look like this Server1:test@Server1 ip-address? |
| 12:57.32 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:57.40 | IsUp | Diffen: yes |
| 12:58.20 | kaldemar | tuxx-: yes. Dial(SIP/username@domain). see the sample sip.conf for more dial strings. |
| 12:58.34 | tuxx- | mkay, tnx :) |
| 12:58.57 | Diffen | IsUp ok hmm im getting wrong password now |
| 12:59.03 | Diffen | thats atleast something new :D |
| 13:01.05 | IsUp | Diffen: may you pastebin your server1 and server2 config again and console output |
| 13:01.55 | Diffen | sure hold |
| 13:09.36 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 13:11.19 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 13:16.12 | Diffen | IsUp: here you go my man: http://pastebin.com/QxqPUQLv |
| 13:18.07 | kaldemar | Diffen: why are you registering btw? |
| 13:18.41 | Diffen | kaldemar dont i need to do that on server 2 to get access to pstn via server1? |
| 13:19.00 | kaldemar | Diffen: no. registerin is just a way to let the other end know where you are. |
| 13:19.19 | Diffen | hmm ok |
| 13:19.25 | kaldemar | that's what authentication is for. but you have disabled it with the insecure option. |
| 13:20.15 | *** join/#asterisk OpenSys (~vasco@fw.vslinux.net) |
| 13:20.18 | OpenSys | hello |
| 13:20.22 | jacc0 | hi |
| 13:20.27 | kaldemar | there's nothing wrong with the configuration in your pastebin, but then again it is fabricating and doesn't show addresses. check and double check the secret again. |
| 13:20.54 | OpenSys | i have a problem with asterisk, is calling so some people automatic |
| 13:20.57 | OpenSys | lol |
| 13:21.12 | OpenSys | have this on lines |
| 13:21.14 | Diffen | kaldemar yes i have changed adress but nothing else. |
| 13:21.14 | OpenSys | Stopping retransmission on |
| 13:21.32 | Diffen | hmm i will try change the password and see if that helps |
| 13:21.35 | OpenSys | Stopping retransmission on '6822b00932b120b628e8c35a71ed07f2@ |
| 13:22.09 | OpenSys | the recalling was disabled |
| 13:22.21 | OpenSys | as i can reset this ? |
| 13:24.43 | [TK]D-Fender | Diffen, your register will fail every time. on the server you are trying tor egister to you have specified a host, not used "dynamic" If you have a fixed host you are forbidden from registering <--- |
| 13:25.49 | Diffen | [TK]D-Fender: Thanks. So if I add dynamic it will work? |
| 13:26.05 | [TK]D-Fender | Well.. it will be allowed to actually try |
| 13:26.23 | [TK]D-Fender | Considering everything else masked and the rest of your environment.. who knows |
| 13:26.48 | Diffen | works perfectly :D |
| 13:26.52 | Diffen | sweet |
| 13:26.59 | Diffen | Thanks for all help guys! |
| 13:28.20 | OpenSys | any ideia about my problem folks ? |
| 13:29.18 | OpenSys | "Stopping retransmission" "Request 102: Match Found" |
| 13:31.26 | [TK]D-Fender | OpenSys, Please pastebin come someplete SIP debug for your problem. The entire transaction, not just some error message |
| 13:31.27 | [TK]D-Fender | ~pb |
| 13:31.28 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:42.33 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 13:48.13 | [TK]D-Fender | some complete* |
| 13:48.14 | [TK]D-Fender | wow.. |
| 13:48.38 | [TK]D-Fender | fried ... guess getting more than 4 hours sleep would have helped.. |
| 13:53.28 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 13:53.41 | leifmadsen | [TK]D-Fender: or lay off the booze :) |
| 13:53.54 | leifmadsen | (not that I encourage that type of behaviour) |
| 13:54.00 | n3hxs | more booze = more sleep? |
| 13:54.09 | [TK]D-Fender | I would... if I drank... late night of music and then attempting to hunt down my lost cell-phone. |
| 13:54.23 | [TK]D-Fender | Going to be a reall shitty day with a small prayer for it getting better |
| 13:58.33 | tzafrir | http://devel.0cpm.org/firmerware/ - open source firmware for some SIP phones (pre-alpha, right now) |
| 14:03.25 | wdoekes2 | something fishy with the verisign whois? $ whois amazon.com | grep AMAZ |
| 14:03.30 | wdoekes2 | <PROTECTED> |
| 14:05.45 | tuxx- | lol wdoekes2 :) |
| 14:05.52 | tuxx- | nice server names ^_^ |
| 14:08.41 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 14:44.31 | *** join/#asterisk irroot (~gregory@41.54.47.36) |
| 14:53.53 | *** join/#asterisk MariusAgon (~MariusAgo@79.142.116.89) |
| 14:57.25 | MariusAgon | Hello. Got a question, found nothing googleing about that. So: if CUSTOMER1 called to PBX1, PBX1 forwarded the call to PBX2, PBX2 connected it with AGENT1. Is there any way for PBX1 to catch DTMF from AGENT1 on PBX2? |
| 14:57.57 | *** part/#asterisk gnude (~gnude@p578bb787.dip0.t-ipconnect.de) |
| 15:02.00 | *** join/#asterisk dmz (~dmz@67.216.138.246.pool.hargray.net) |
| 15:03.07 | [TK]D-Fender | MariusAgon, Depends on your definition of "forwarded" |
| 15:03.28 | [TK]D-Fender | MariusAgon, In general Every leg can see DTMF pass along the way |
| 15:06.15 | MariusAgon | Thank you alot. |
| 15:06.57 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:15.08 | phl4kx | hi all |
| 15:16.27 | *** join/#asterisk OldMonk (~raju@182.68.215.125) |
| 15:17.53 | OldMonk | hi. i'm using Set(CHANNEL(musicclass)=my-class) before invoking Queue; however, the queue is playing MOH class default. any hints? |
| 15:19.08 | irroot | OldMonk there is a setting in queues.conf for MOH |
| 15:19.22 | OldMonk | can't i override that through the dialplan? |
| 15:19.50 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:20.12 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
| 15:20.19 | irroot | musicclass |
| 15:20.31 | OldMonk | irroot: that's what i |
| 15:20.36 | OldMonk | 'm trying to do :) |
| 15:21.23 | irroot | the queue config has a musicclass setting too |
| 15:21.43 | OldMonk | irroot: and i'm trying to override that through the dialplan |
| 15:22.19 | OldMonk | in fact, it isn't set at all in queues.conf |
| 15:23.06 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:23.14 | irroot | im looking at source to figgure it out |
| 15:25.04 | OldMonk | great |
| 15:25.18 | irroot | OldMonk looks like it is set from queues.conf setting only |
| 15:25.47 | OldMonk | oh, it's mixing default and my-class! |
| 15:25.54 | irroot | if itsnot set i suspect it defaults to default |
| 15:26.09 | OldMonk | that's the most weird thing i ever heard! |
| 15:26.20 | irroot | so if you add a musicclass setting to queue in queues.conf then it will work |
| 15:26.31 | OldMonk | i need a dynamic musicclass |
| 15:26.51 | irroot | <PROTECTED> |
| 15:27.07 | [TK]D-Fender | <OldMonk> irroot: and i'm trying to override that through the dialplan <- not happening |
| 15:27.14 | irroot | OldMonk please log it as a jira issue and mark me in it |
| 15:27.38 | OldMonk | [TK]D-Fender: er? |
| 15:27.46 | [TK]D-Fender | OldMonk, Stop setting the MoH class int eh queue, make sure that the general settings in queues.con also leaves it out and it should take the channel's mode. Set it yourself all the time |
| 15:28.19 | [TK]D-Fender | Channel MOH should not override a queues.conf MoH setting. |
| 15:28.34 | OldMonk | [TK]D-Fender: that's exactly what i'm doing |
| 15:28.57 | [TK]D-Fender | OldMonk, If you specify it then you can't override it |
| 15:28.58 | irroot | [TK]D-Fender yeah if its set in genral it could come into play perhaps i dont see it as genral option in the source |
| 15:29.46 | OldMonk | [TK]D-Fender: it's not specified at all in queues.conf. 20:52 OldMonk> in fact, it isn't set at all in queues.conf |
| 15:30.15 | OldMonk | it's bloody mixing the default moh and my-class! how can that happen? |
| 15:30.45 | OldMonk | first default plays, then the two get mixed, then default stops and just my-class plays |
| 15:32.09 | *** join/#asterisk timahvo1 (~rogue@41.90.8.237) |
| 15:32.48 | [TK]D-Fender | ~pb |
| 15:32.48 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:32.50 | [TK]D-Fender | ^^^ |
| 15:34.43 | irroot | <PROTECTED> |
| 15:35.31 | irroot | the class set needs to be valid however |
| 15:35.49 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 15:36.06 | OldMonk | ah wait, there's no mixing, my client was confused |
| 15:37.23 | *** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:44.31 | OldMonk | sorry, my bad: the files were put into my-class after asterisk restart, and asterisk didn't recognise the empty directory as a valid music class |
| 15:45.02 | OldMonk | asterisk restart has fixed the problem |
| 15:45.12 | OldMonk | thanks irroot, [TK]D-Fender |
| 15:45.48 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:53.17 | OldMonk | later |
| 15:57.38 | *** part/#asterisk RobertLaptop (~RobertLap@pool-173-69-205-54.bltmmd.fios.verizon.net) |
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| 15:58.38 | *** join/#asterisk navaismo (~navaismo@189.146.58.40) |
| 16:02.56 | *** part/#asterisk asterisk-Tester (~Ramyt@193.227.170.247) |
| 16:04.40 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 16:04.50 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 16:05.16 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 16:05.36 | sereal-work | what aspects of sip.conf relate to call transfer? (I'm doing call transfer via the phones, not threw a exten) |
| 16:06.57 | sereal-work | I have multiple companies running on the same asterisk, one company their call transfer works, other other doesn't. |
| 16:07.11 | sereal-work | I can only guess the problem lies in something I left out in the sip.conf |
| 16:07.20 | sereal-work | or *maybe* it has something to do with the phone configurations? |
| 16:10.31 | SeRi | sereal-work, I might be wrong but call transfer has nothing to do with sip.conf. sip.conf is only to set your peers/device maybe your dial plan could have something wrong. or your peers do not have a context define correctly, etc.... |
| 16:11.29 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 16:12.42 | phl4kx | hi all |
| 16:12.48 | phl4kx | I have this error in my asteerisk |
| 16:12.56 | phl4kx | [Nov 25 06:09:41] NOTICE[1177]: chan_dahdi.c:3059 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 |
| 16:12.58 | phl4kx | some help please |
| 16:13.54 | sereal-work | SeRi, I think that is correct. Their contexts should be defined fine, what happens is I hit the transfer button and it dials the other person but the call is never actually transfered over. |
| 16:14.08 | phl4kx | is a TE122B connect to CLARO PERU |
| 16:17.02 | MrTelephone | Do you guys ever have issues with other phone ivr's not recognizing digits from asterisk over a pri? |
| 16:17.16 | MrTelephone | I'm wondering what setting I can try to fix this. Some phone switches work and others don't |
| 16:17.58 | navaismo | sereal-work: pb your sip debug of failed transfer |
| 16:18.35 | sereal-work | pb? |
| 16:18.44 | navaismo | ~pb |
| 16:18.44 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:18.50 | sereal-work | ohh sorry. |
| 16:19.13 | navaismo | sorry i dont send you before |
| 16:20.22 | sereal-work | one sec I gotta get my test equipment up and running |
| 16:25.30 | navaismo | phl4kx: what other message do you see, yellow alarm maybe? |
| 16:27.33 | navaismo | MrTelephone: maybe the dtmf in the phone its wrong usually is rfc2833 |
| 16:28.20 | MrTelephone | i'll check it out. I think the sangoma card screws with the dtmf. Maybe I have a setting wrong |
| 16:29.16 | MrTelephone | here there is a problem with the voicemail_users table for asterisk 1.6+. The password column is INT(4) which means you can't have a leading zero for a password. How can I fix this without breaking anything? Would it still work with VARCHAR data tybe? |
| 16:29.20 | FlashDeluxe | sereal-work: what phones do you use? |
| 16:31.46 | sereal-work | aastra 6731i |
| 16:31.58 | navaismo | MrTelephone: we use it with varchar(11) |
| 16:31.58 | sereal-work | and the aastra non poe ones |
| 16:32.08 | FlashDeluxe | sereal-work: i guess you forgot to activate the option RTP (RFC2833) enalbe or sth. like this in the phones? |
| 16:32.27 | MrTelephone | navaismo, alright thanks for the heads up |
| 16:32.29 | sereal-work | humm I didn't think of that - that could be the case. |
| 16:34.02 | FlashDeluxe | sereal-work: the whole transfer stuff is configured in /etc/asterisk/features.conf if you don`t send DTMF stuff it won`t work :) |
| 16:34.44 | sereal-work | thanks. That might be the problem I over looked. features.conf should be setup correctly as in another context transfers work. |
| 16:35.17 | FlashDeluxe | sereal-work: np |
| 16:38.26 | *** join/#asterisk jpcansa (~JP@190.211.93.11) |
| 16:41.33 | jpcansa | hi, is there any way to check the blf state of the phones from within the CLI? |
| 16:41.53 | navaismo | try with core show hints |
| 16:42.06 | SeRi | ^^ |
| 16:43.40 | jpcansa | navaismo, great, thats what i needed |
| 16:45.28 | navaismo | jpcansa: no problem |
| 16:45.44 | jpcansa | why would a phone stays in "State:inuse" even if the phone is not being used, I even restarted both the operartor phone with the BLF console and the end user phone |
| 16:46.46 | *** join/#asterisk scubes13 (~scubes13@66-191-209-216.dhcp.spbg.sc.charter.com) |
| 16:49.42 | phl4kx | navaismo all is OK |
| 16:49.44 | phl4kx | no more message |
| 16:49.48 | phl4kx | the LED is GREEN |
| 16:50.17 | phl4kx | but when i make a call I have noise during call |
| 16:50.29 | phl4kx | and sometimes my channels are restarted |
| 16:51.01 | navaismo | maybe you are having issues with the clock |
| 16:51.14 | phl4kx | with clock? |
| 16:52.05 | navaismo | yes let find the propper term |
| 16:53.18 | navaismo | the "synchronization with the e1" |
| 16:58.41 | phl4kx | hahaha |
| 16:58.43 | phl4kx | ok |
| 17:05.32 | MrTelephone | relaxdtmf=yes but I have hardware DTMF. maybe that is my problem |
| 17:14.13 | MrTelephone | do you guys have with digital e1s/t1s ever have to play around with your tx/rx gains? |
| 17:15.06 | anonymouz666 | never |
| 17:15.44 | navaismo | never |
| 17:17.20 | MrTelephone | man my english is bad |
| 17:17.34 | navaismo | mine too |
| 17:17.42 | MrTelephone | sorry, running on like 4 hours of sleep :( |
| 17:18.08 | anonymouz666 | MrTelephone: don't worry, english is not my native language :P |
| 17:18.11 | MrTelephone | when I get halfway through a sentence I don't even know what the first half says |
| 17:18.38 | MrTelephone | I think it's from staring at programming languages all the time |
| 17:19.05 | MrTelephone | what is your native language? |
| 17:19.51 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 17:19.52 | MrTelephone | I think the sangoma card is messing with the DTMF because internall dtmf works good. I have all the clients set to rfc2833. |
| 17:21.05 | MrTelephone | I still don't understand where all the echo comes from when you use a t1/e1. The convergence to the IP network causes that much delay? If you hook up a channel bank to a t1 with some analog phones on it there is no echo. |
| 17:22.54 | MrTelephone | I don't know much about echo and timing. At what milliseconds do we start to hear echo? |
| 17:23.22 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:26.03 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 17:28.22 | navaismo | MrTelephone: http://kb.digium.com/articles/Troubleshooting/General-Echo-HOWTO?retURL=%2Fapex%2FknowledgeProduct%3Fc%3DDigital&popup=false |
| 17:29.58 | akrohn | this is probably a silly question, but what conditions need to be met in order to execute the timeout extension? I'm rewriting all of our existing dialplans from scratch as I learn. I have the 't' extension playing 'vm-goodbye' and then hanging up. but after greeting plays, the call simply ends without playing the goodbye message. any ideas? |
| 17:31.14 | MrTelephone | is your priority right? |
| 17:31.22 | MrTelephone | t,1,playback(goodbye) |
| 17:31.26 | MrTelephone | t,1,Hangup() |
| 17:31.31 | MrTelephone | oops |
| 17:31.34 | MrTelephone | t,n,Hangup() |
| 17:32.23 | akrohn | yessir. i have exten => t,1,Playback(vm-goodbye); same => n,Hangup() |
| 17:32.31 | MrTelephone | navaismo, I have that expensive echo cancelling but it still sometimes makes the calls sound half-duplex |
| 17:33.46 | MrTelephone | what does the console say, nothing? |
| 17:34.15 | MrTelephone | you called watexten() and didn't press nothing? |
| 17:34.23 | akrohn | that's correct |
| 17:34.49 | MrTelephone | I checked my dialplan and I have a macro there so it is just a one-liner |
| 17:35.10 | [TK]D-Fender | akrohn, pastebin your acomplete dialplan and the failed call from CLI |
| 17:35.12 | MrTelephone | maybe timeout doesn't allow priority? I can't see that though |
| 17:35.13 | navaismo | timeout is executed when you never press a digit to select extension, i.e., in and IVR you set your background your waitexten your menu and the t exten if the user never make a choice when the waitexten reach the time it will jump to the timeout |
| 17:35.31 | navaismo | note that you need to answer() before playing messages |
| 17:35.48 | [TK]D-Fender | lNo |
| 17:35.56 | navaismo | :( |
| 17:35.57 | akrohn | kk one sec |
| 17:35.59 | navaismo | LOL |
| 17:36.04 | [TK]D-Fender | Playback Answer's if you haven't already explicitly done so. |
| 17:36.51 | MrTelephone | can I jump to asterisk 10.0.0 from 1.8.5 or am I digging a grave? |
| 17:37.32 | dijib | anybody familier with tc & iptables, could you take a min to error check http://pastebin.com/XQhhBjcH |
| 17:37.36 | dijib | ?? |
| 17:37.45 | dijib | and im now using screen and irssi so i rock |
| 17:37.50 | *** join/#asterisk brainiac (~brainiac@24-183-217-180.dhcp.jcsn.tn.charter.com) |
| 17:38.01 | akrohn | http://pastebin.com/1sGtFyrL |
| 17:40.09 | MrTelephone | Do you have to have a hangup clause in there, is it screwing things up |
| 17:40.21 | MrTelephone | I think your extension is erroring out because it doesn't like that clause |
| 17:40.31 | MrTelephone | and by default it ends the call |
| 17:41.00 | MrTelephone | I guess it just says warning tho |
| 17:41.03 | MrTelephone | :-/ |
| 17:41.21 | akrohn | i'll take the cause codes out and see if that changes it |
| 17:41.59 | MrTelephone | where is your t, does that pastebin take stuff out? |
| 17:42.10 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 17:42.12 | akrohn | at the very bottom |
| 17:42.25 | akrohn | line 54 |
| 17:43.01 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 17:43.06 | MrTelephone | does your invalid work? |
| 17:43.18 | akrohn | yes |
| 17:43.31 | dijib | why does #lartc suck so bad |
| 17:43.35 | dijib | im going for a smoke |
| 17:43.37 | [TK]D-Fender | akrohn, Timeout doesn't because there is a prioirty after your waitexten |
| 17:43.53 | [TK]D-Fender | akrohn, Remove it |
| 17:44.01 | akrohn | ooohhhhhhhh |
| 17:44.05 | akrohn | genius |
| 17:44.24 | akrohn | so if i make sure to hangup everywhere else i shouldn't need the one after waitexten |
| 17:46.02 | MrTelephone | the call will hangup regardless if there is nowhere else for the call to go |
| 17:46.16 | akrohn | thanks for the help. troubleshooting lesson learned |
| 17:46.40 | *** join/#asterisk dhorner_mb (~dhorner_m@184.18.44.244) |
| 17:46.57 | akrohn | asterisk is mindblowingly useful and feature-rich. i've been supporting our current setup for about 6 months and think i can start working on building a new one from scratch. obviously i have a lot to learn though |
| 17:47.38 | MrTelephone | it's brutal |
| 17:47.55 | MrTelephone | fender has a good eye though. I didn't catch your mistake |
| 17:48.30 | akrohn | indeed. i'm a programmer and that's one of those things that's very easy to overlook |
| 17:50.42 | akrohn | worked like a charm. you guys rock. i should start hanging out here more |
| 17:51.47 | SeRi | dijib, you still at it? |
| 17:55.31 | SeRi | dijib, Again I really think you should go for the native QoS/Shaping of openwrt before you try to implement tc. Why are you so bent on using this script? The native openwrt QoS/Shaping also uses tc. go to the openwrt wiki and read all about it.... it is all over their wiki..... |
| 17:56.53 | SeRi | if you must have tc google openwrt tc qos you will get a bunch of hits with examples made for openwrt.... |
| 17:57.07 | SeRi | dijib, http://wiki.openwrt.org/doc/howto/tc/tc.theory |
| 18:00.12 | MrTelephone | akrohn, the dialplan becomes unbearable after a while I'll tell you that |
| 18:00.18 | MrTelephone | I'm up to 800 lines now |
| 18:01.02 | MrTelephone | Spaghetti? |
| 18:01.29 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:02.10 | akrohn | well MrTelephone I was wondering about that as well. we have all our customers sorted out into their own config files and use a lot of includes. do you have any advice on how to best organize the dialplan code and minimize repetition? as it is, we include the same chunk of code once or twice per customer and seems excessive to repeat it in all the contexts |
| 18:02.41 | [TK]D-Fender | Macros <- |
| 18:03.18 | akrohn | macros are going to save my life, huh? |
| 18:03.20 | [TK]D-Fender | MrTelephone, I'd be interested in seeing what you've done with 800 lines... |
| 18:03.46 | [TK]D-Fender | akrohn, Saves on repetition. |
| 18:03.59 | IsUp | [TK]D-Fender: you should see my servers, 1694 lines exactly :> |
| 18:04.09 | [TK]D-Fender | Share away.... |
| 18:04.24 | MrTelephone | I have a lot of repetition |
| 18:04.29 | MrTelephone | nothing overly complicated |
| 18:04.45 | [TK]D-Fender | MrTelephone, PEBKAC :) |
| 18:04.51 | MrTelephone | I have to handle 6 offices and 120 residential |
| 18:05.00 | MrTelephone | so there are dialplan rules for each |
| 18:05.09 | IsUp | same here. we have SIP trunks, 3 x PRI and 1 SS7 link. |
| 18:05.32 | IsUp | real spaghetti |
| 18:07.10 | MrTelephone | how did you negotiate your ss7 link? |
| 18:07.24 | MrTelephone | did you have to get lawyers and all that? |
| 18:07.51 | IsUp | MrTelephone: chan_ss7 with 1.4 :p |
| 18:08.38 | MrTelephone | I don't have a provider that does ss7 that I know of |
| 18:08.46 | MrTelephone | maybe if I get into some sord of trunk agreement |
| 18:09.18 | *** join/#asterisk seather (~seather@tyrmida.com) |
| 18:10.04 | IsUp | MrTelephone: i am working on a VAS company, we are hosting adult lines and telemarketing stuff |
| 18:10.06 | MrTelephone | akrohn, i'm trying to synchronize asterisk with our management database so I'm changing the dialplan to do ODBC lookups. So outgoing calls will check for suspended='1' and if it is the call won't go through |
| 18:10.21 | IsUp | MrTelephone: all GSM operators are requesting SS7 |
| 18:11.06 | seather | Evening, when the internet goes down and comes back up, my sip peers and registrations fail to come back online again, weven when issuing sip reload peers/registry command. I basically have to reboot the entire box to get things working again. Is this a known problem or is there anything I can try? |
| 18:11.07 | MrTelephone | I see. I read a little bit about SS7. Is that just a differnet protocol on the same hardware. T1/E1/T3/etc. |
| 18:11.41 | MrTelephone | seather, is your ip address changing? |
| 18:11.57 | seather | MrTelephone: yes, it is a dynamic nat'ed connection |
| 18:12.15 | SeRi | seather, what distro? asterisk version? |
| 18:12.23 | IsUp | MrTelephone: yeah exactly, we are using Sangoma card. it works with T1/E1 |
| 18:12.34 | MrTelephone | akrohn, right now I have to change the clients context manually in order to disconnect them and then do a reload. I don't like doing reloads because sometimes it f's up the registrations |
| 18:13.07 | SeRi | seather, There was a known bug on earlier version of asterisk 1.8.x that peers would not re register once they would loose want connectivity... |
| 18:13.15 | seather | SeRi: oldsh 1.6.0.26 (Trixbox CE) |
| 18:13.24 | MrTelephone | Our network has outages too and I don't have that problem seather. I don't see why you have to reboot, what if you just restart asterisk? |
| 18:13.31 | SeRi | s/wnats/wan |
| 18:13.33 | seather | think this one is trixbox 2.8 |
| 18:13.37 | SeRi | s/want/wan |
| 18:13.51 | SeRi | seather, head to #trixbox for help |
| 18:14.00 | SeRi | ~trixbox |
| 18:14.01 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 18:14.09 | seather | yeah I am ware of that |
| 18:14.15 | seather | was just wondering if anyone is aware of a bug |
| 18:14.22 | seather | thanks SeRi |
| 18:14.32 | seather | I'll go find the bug |
| 18:14.45 | SeRi | seather, It would be hard to determine if the bugs is in trixbox ro asterisk :( |
| 18:14.54 | SeRi | s/ro/or |
| 18:14.59 | SeRi | s/ro/or/ |
| 18:15.03 | SeRi | lmao |
| 18:15.06 | SeRi | damn boot! |
| 18:15.06 | seather | well if I at least know what the bug in asterisk was I can go see if it is the same problem I am having :) |
| 18:16.14 | MrTelephone | I finally got around to doing an incoming blacklist for customers so they can block bill collectors and telemarketers |
| 18:16.37 | [TK]D-Fender | seather,You aren't on the latest release of your branch nor on one that is currently supported |
| 18:18.24 | *** join/#asterisk adeel|work (~adeel@unassigned-220.80.183.216.net.blink.ca) |
| 18:19.32 | *** join/#asterisk qakhan (~qakhan@180.178.175.19) |
| 18:20.03 | qakhan | can someone tell me voice recognition in asterisl |
| 18:20.08 | qakhan | asterisk* |
| 18:21.01 | *** join/#asterisk happylife (~happylife@46.251.83.126) |
| 18:21.35 | [TK]D-Fender | qakhan, Sphinx = meh + free. Lumenvox = not bad + affordable |
| 18:22.24 | qakhan | Sphinx = meh + free??? |
| 18:22.46 | Gugge | its not great, and free |
| 18:23.15 | qakhan | sphinx is not good? |
| 18:23.26 | Gugge | its "meh" :) |
| 18:23.54 | qakhan | sorry i didnt get u |
| 18:23.57 | MrTelephone | macros return to the previous context. Is there a way to stop that. Hangup()? |
| 18:28.09 | [TK]D-Fender | MrTelephone, Certainly |
| 18:38.10 | qakhan | plz tell about sphinx? |
| 18:39.31 | [TK]D-Fender | What is there to tell? Quality = not so great. Google up some guides and go try it |
| 18:41.31 | qakhan | i didnt find on google |
| 18:41.38 | qakhan | cau u help me? |
| 18:44.34 | [TK]D-Fender | http://www.google.ca/#hl=en&cp=21&gs_id=3q&xhr=t&q=asterisk+sphinx+howto&pf=p&sclient=psy-ab&source=hp&pbx=1&oq=asterisk+sphinx+howto&aq=f&aqi=&aql=&gs_sm=&gs_upl=&bav=on.2,or.r_gc.r_pw.r_cp.,cf.osb&fp=76f0411462f461ec&biw=1600&bih=927 |
| 18:44.45 | [TK]D-Fender | Funny I see a full page right there |
| 18:49.36 | *** join/#asterisk moy (~moy@190.191.218.85) |
| 18:50.18 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
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| 19:01.15 | adeel|work | i have a pretty hefty box which is running * and i'm trying to get it to scale nicely...so far, using SIP-SIP with only g711, i can only seem to push through about 140-200 concurrent calls...any tips on what to tweak to try bumping up that number? |
| 19:01.18 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 19:02.05 | ChannelZ | What are you running out of? CPU? Bandwidth? |
| 19:02.30 | adeel|work | ChannelZ, nothing...it's on a full GigE link, and cpu is not above 20% |
| 19:02.47 | adeel|work | load is 0.36 0.21 0.11 |
| 19:03.15 | adeel|work | i've tried setting all the ulimit params to unlimited |
| 19:03.50 | [TK]D-Fender | adeel|work, pastebin a typical call. |
| 19:04.07 | adeel|work | one sec |
| 19:04.45 | adeel|work | oh, and i'm using sipp to generate the load with a UAC with media Profile on 1 ext, and the other running the UAS with media profile to simulate a full duplex call |
| 19:06.03 | [TK]D-Fender | adeel|work, make sure that you aren't actually overloading the tester as well.. hopefull that is on a separate beefy box. |
| 19:08.10 | adeel|work | the tester is a quad-core box with 16gb ram with gigE link...so it shouldn't be too loaded |
| 19:10.27 | *** join/#asterisk niekie (quasselcor@CAcert/Assurer/niekie) |
| 19:11.28 | [TK]D-Fender | adeel|work, A separate box from the one you're testing it with, right? |
| 19:12.02 | adeel|work | yeah |
| 19:12.51 | adeel|work | [TK]D-Fender, http://pastebin.com/1YJ1i3Aa |
| 19:13.53 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-votnmfhjraurpbpi) |
| 19:15.11 | [TK]D-Fender | VNDial |
| 19:15.18 | adeel|work | i can't really mess with the dialplan though |
| 19:15.23 | [TK]D-Fender | Ok, yeah I'm seeing a who pile of customized looking dialplan apps |
| 19:15.44 | [TK]D-Fender | Aside from the fact that the basic readable diaplan is complex... its these odd apps that make me wonder what we're looking at |
| 19:16.00 | [TK]D-Fender | What is the core * under there and where is all this custom stuff from? |
| 19:16.20 | adeel|work | custom modules we did in house for our setup |
| 19:16.25 | adeel|work | the core is 1.6.2 |
| 19:16.27 | adeel|work | er, 20 |
| 19:17.03 | adeel|work | it's a realtime setup |
| 19:17.40 | [TK]D-Fender | ok, harder to judge from that... here's an idea : Tt <- remove the dialplan options that force * to stay in the path and process live |
| 19:18.06 | [TK]D-Fender | Not saying start forcing reinvites, just stop * from having to decode rather than repack. |
| 19:18.14 | [TK]D-Fender | See how that impacts your limits |
| 19:18.46 | adeel|work | ah, see if it's the dialplan that's limiting it or if its an */OS limitation |
| 19:19.02 | adeel|work | are there any other OS level optimizations i can do before fiddling with the dialplan? |
| 19:20.09 | [TK]D-Fender | I'm not that technically minded... |
| 19:22.15 | adeel|work | hmmm....ok |
| 19:22.17 | adeel|work | thanks |
| 19:28.17 | SeRi | adeel|work, I might be wrong but asterisk optimization is base on CPU/RAM not much else you can tweak on the OS to make asterisk scale.... Only thing that comes to mind are limits and priority's on the OS but looking at your post looks like when you are reaching 200 calls the system does not seem stressed so to me it seem you have a bottle neck somewhere else.... (n00b here) |
| 19:30.40 | adeel|work | SeRi, yeah, but now the fun begins in trying to track down where that bottleneck could occur.... |
| 19:31.26 | SeRi | adeel|work, welcome to the sys admin's world :P |
| 19:31.39 | adeel|work | let me try the same tests without any media, and see what's up |
| 19:31.54 | adeel|work | SeRi, heh, i'm aware been sysadmining for a while now |
| 19:31.56 | [TK]D-Fender | First = media only, not listening in |
| 19:32.37 | adeel|work | i ran the no media test, and got a nice call clearing rate; but i hadn't verified the actual test scenario |
| 19:32.54 | adeel|work | i'm going to try repeating it and see if i hit the numbers i got yesterday |
| 19:34.13 | SeRi | adeel|work, If you are not passing video or images I would disable it. you should strip asterisk down to what you need only. unload all those modules that are not need it by your dial plan..... |
| 19:34.20 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 19:34.34 | adeel|work | yeah, i'll be doing that too shortly |
| 19:34.39 | SeRi | strip modules, applications, etc.... |
| 19:35.17 | SeRi | bbl |
| 19:35.22 | SeRi | good luck |
| 19:44.24 | dijib | anybody know how to review lines that have gone off screen in irssi? |
| 19:45.00 | dijib | and good day SeRi |
| 19:48.31 | hardwire | page up? |
| 19:48.48 | hardwire | don't forget page down once you're up there. |
| 19:48.54 | hardwire | I HOPE IT'S NOT TOO LATE! |
| 19:56.45 | dijib | after locating the pgup & pgdwn buttons on this KB works like a charm |
| 19:57.29 | anonymouz666 | people calling friday 6pm, could be only to wish "good weekend" |
| 19:58.03 | anonymouz666 | :P |
| 20:16.02 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 20:20.49 | n3hxs | anonymouz666, my luck it is to handle issues that have been sitting on their desk since Wednesday. |
| 20:51.40 | Greenlight | Do the cdr record in asterisk 10 still go into mysql by default? |
| 20:56.02 | ChannelZ | hmm they've never gone into mysql "by default" |
| 20:56.50 | Greenlight | Na, I mean like I recall reading about asterisk 10 using SQLite, rather than mySQL |
| 20:57.04 | Greenlight | Do I need to do anything funky to keep them in mysql |
| 20:58.56 | ChannelZ | presumably not if you're configured that way |
| 21:00.08 | Greenlight | Well I'm doing a fresh install, my last was an upgrade. Guess I'll soon find out :) |
| 21:02.34 | *** join/#asterisk master_of_master (~master_of@p57B5447C.dip.t-dialin.net) |
| 21:05.54 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 21:05.54 | *** mode/#asterisk [+o file] by ChanServ |
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| 21:41.37 | kaldemar | Greenlight: nothing uses mysql by default. asterisk uses sqlite instead of berkley DB as an inner database. if you want to use mysql, you must configure its usage, as always. |
| 21:41.54 | dijib | http://i.imgur.com/ZQJAQ.jpg |
| 21:43.03 | Greenlight | Ahh right so it's just the inner database, and nothing else that's been changed. I usually just edited cdr_mysql.conf and set it up that way, but when I did menuselect on asterisk 10 it seemed like that module had been depreched |
| 21:44.39 | GG111 | Anyone has experience with making Google voice calls coming in successfully to his Asterisk >? |
| 21:50.22 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:50.24 | [sr] | blue |
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| 22:07.49 | GG111 | No Google voice advice :) ? |
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| 22:12.35 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
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| 22:27.01 | leifmadsen | tzanger: ping |
| 22:27.38 | leifmadsen | GG111: it's a holiday in the US so there is likely very little traffic. To answer your question, "yes, people have successfully made calls using google voice" |
| 22:28.10 | tzanger | leifmadsen: pong |
| 22:28.12 | leifmadsen | one problem is that google keeps flipping servers around, so the authentication methods tend to work and then not work, and then work again. You can search the bug tracker for the patches that keep getting applied to fix it. |
| 22:28.25 | leifmadsen | tzanger: so I'm very close to having an automated LDAP installation on an ubuntu server |
| 22:28.38 | tzanger | what are you using the ldap for, precisely? |
| 22:28.50 | leifmadsen | egroupware, and then asterisk integration |
| 22:29.02 | tzanger | interesting |
| 22:29.06 | neurosys | leifmadsen, Running the latest asterisk-gui SVN, but for some reason, its not writing the trunks to sip.conf, or any other options. They wont save. I see nothing in the logs or debug. any thoughts? |
| 22:29.16 | leifmadsen | neurosys: none whatsoever |
| 22:29.17 | tzanger | I've foudn that ldap seems to be so poorly documented because it's very implementation-dependent |
| 22:29.33 | neurosys | leifmadsen: :( |
| 22:29.34 | leifmadsen | tzanger: ya for sure... I've made some really good strides over the last 2 days |
| 22:29.40 | leifmadsen | neurosys: never used the asterisk-gui |
| 22:29.48 | tzanger | how did you decide on which schema to use? |
| 22:29.54 | leifmadsen | that's the trick I guess :) |
| 22:30.06 | neurosys | leifmadsen, who's heading up that dev? |
| 22:30.42 | leifmadsen | tzanger: I have a very basic schema to start, and the egroupware seems to be able to take over from there. For Asterisk the schema is in the contrib/scripts/ directory |
| 22:30.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:30.57 | tzanger | hmm |
| 22:31.10 | leifmadsen | tzanger: once you get things going, then things like phpldapadmin seem to be fairly useful for looking at the current schema and modifying it |
| 22:31.15 | leifmadsen | tzanger: this got me started: https://help.ubuntu.com/10.04/serverguide/C/openldap-server.html |
| 22:31.20 | tzanger | interesting, ok |
| 22:31.45 | leifmadsen | tzanger: I've pieced together other things, but that page is the only one I can find that shows how to start with openldap using the slapd.d method (cn=config they call it) |
| 22:32.04 | leifmadsen | everything else is slapd.conf which seems to be outdated (although I figured out how to make ubuntu go back to slapd.conf) |
| 22:32.08 | leifmadsen | neurosys: no one currently |
| 22:32.09 | dijib | SeRi: this is for you http://pastebin.com/WkXfbfwa |
| 22:32.17 | neurosys | leifmadsen, roger that |
| 22:32.19 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 22:33.49 | SeRi | dijib, you ihn? |
| 22:33.54 | SeRi | s/inh/in |
| 22:34.03 | SeRi | stupid boot |
| 22:37.13 | *** join/#asterisk jpcansa (~JP@190.211.93.11) |
| 22:38.16 | *** join/#asterisk jercos (jercos@gateway/shell/anapnea.net/x-aagjknpkzyrmyyla) |
| 22:38.25 | SeRi | dijib, what do you want me to do with that? |
| 22:38.55 | jercos | Howdy folks. So I'm trying to run DISA via manager directly putting a dialed user into a given context ready to dial... |
| 22:39.36 | jercos | So I give action originate, channel as a sip channel that rings, Application: DISA, and Data: no-password|DLPN_LocalAnalog |
| 22:40.07 | jercos | DLPN_LocalAnalog being the context used for a set of Zap FXS lines >.> |
| 22:40.21 | jercos | And yet when the phone rings, and I answer, I get a dialtone, but it appears to be in the default context. |
| 22:40.52 | leifmadsen | you'll need to show what you're doing with manager and what the console output looks like, along with the dialplan configuration |
| 22:41.57 | jercos | I showed what I was doing with manager, I'm logged into manager, and I paste http://pastebin.com/tUmVazaB |
| 22:42.14 | jercos | Hole on a sec, I'll go fetch the console output from doing that |
| 22:42.44 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-vaeirluhoxwmcbga) |
| 22:44.22 | jercos | Blah, information everywhere. Whatever, done with SIP/6010, a registered softphone. Same results. |
| 22:44.25 | jercos | http://pastebin.com/yU1DpyaF |
| 22:44.34 | jercos | That;s from manager |
| 22:44.36 | tzanger | one thing I don't like about ubuntu is how they try to take things out of the config files and put them in their /etc/default/blah scripts |
| 22:44.45 | tzanger | leifmadsen: thanks for the update, I appreciate it |
| 22:44.52 | jercos | Is there something particularly special I'll need from the console that manager doesn't show? |
| 22:45.46 | leifmadsen | tzanger: ya that's true, although I had to make very few changes to /etc/default/slapd in this case. Been working on a Puppet module to do automated asterisk (and other software) installs. Been about a day and a half, and the end is in site for the openldap module. |
| 22:47.49 | dijib | can someone offer to me an explaination as to what in this if statement is broken |
| 22:47.52 | dijib | http://pastebin.com/Jd0t6ir4 |
| 22:48.00 | SeRi | dijib, hey |
| 22:48.06 | dijib | yo |
| 22:48.09 | leifmadsen | in what language? |
| 22:48.11 | jercos | dijib: the lack of spaces in the [] |
| 22:48.16 | dijib | so i figured out why most of the errors are happening |
| 22:48.22 | dijib | lack of spaces ya? |
| 22:48.26 | jercos | [ "${1}" = "status" ] |
| 22:48.27 | jercos | ? |
| 22:48.28 | IsUp | 1.if [ "${1}"="status" ] |
| 22:48.31 | leifmadsen | probably a missing trailing ; |
| 22:48.45 | leifmadsen | throws out wild guesses since no language has been defined |
| 22:48.49 | jercos | the newline eliminates the need for a semicolon iirc |
| 22:48.58 | leifmadsen | I'm assuming he means bash |
| 22:49.01 | leifmadsen | jercos: you're probably right |
| 22:49.17 | dijib | like i know the language |
| 22:49.25 | SeRi | dijib, spaces. you in the chan? |
| 22:49.29 | dijib | #!/bin/bash |
| 22:49.35 | dijib | will be in 2mins if your inbound |
| 22:49.51 | SeRi | im in |
| 22:50.06 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 22:50.31 | SeRi | dijib, brb in 5 min. |
| 22:50.43 | dijib | k |
| 22:54.59 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-wmhnybvoytaaujao) |
| 22:56.18 | tzanger | leifmadsen: I'm dumb... what's puppet? |
| 22:56.27 | leifmadsen | tzanger: oh snap |
| 22:56.31 | leifmadsen | www.puppetlabs.com |
| 22:56.39 | tzanger | I have to write a SIP autoreg dialplan on the plane this weekend |
| 22:57.24 | leifmadsen | tzanger: basically an application for scripting automated installations |
| 22:58.12 | leifmadsen | tzanger: if you decide to get into it, do yourself a favour and by the Pro Puppet book -- it's very good |
| 22:58.21 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 22:58.21 | *** mode/#asterisk [+o file] by ChanServ |
| 23:01.28 | tzanger | leifmadsen: interesting |
| 23:01.59 | jercos | Not just installations though, config changes on a mass scale too :3 |
| 23:02.03 | jercos | 'svery handy |
| 23:02.03 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1279682108.dsl.bell.ca) |
| 23:02.08 | leifmadsen | jercos: +1 |
| 23:02.08 | jercos | Keeps everyhting in sync |
| 23:02.20 | leifmadsen | yep, great if you're going to be managing multiple systems |
| 23:02.35 | leifmadsen | I've just started doing templating for the openldap module, and that's incredibly handy as well |
| 23:02.50 | jercos | Absolutely essential if you're going to be managing multiple hundreds of systems :) |
| 23:11.53 | leifmadsen | which everyone will be since everything is now IN THE CLOUD! |
| 23:11.57 | leifmadsen | file: THE CLOUD RIGHT?! |
| 23:15.02 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 23:18.53 | dijib | SeRi: http://pastebin.com/ywbd0rwQ |
| 23:21.27 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-aonsglqgqmfspdtd) |
| 23:31.29 | SeRi | https://forum.openwrt.org/viewtopic.php?pid=147663 |
| 23:31.43 | SeRi | http://www.google.com/search?gcx=c&sourceid=chrome&ie=UTF-8&q=tc+qdisc+dev+imq0+the+root+2#pq=tc+qdisc+dev+imq0+the+root+2&hl=en&sugexp=ernk_spiketing&cp=4&gs_id=k&xhr=t&q=openwrt+qos+tc&qe=b3Blbg&qesig=9GKIil-MN_36k0fzegn38g&pkc=AFgZ2tmXwfHHz3YRfDpQiNSccSa5BS694Hax-Y_KosqEK4107TWTHLxSkFf3r7UrELNdTbo0UpmKWYl4nfDLzpmAGlRydrB12A&pf=p&sclient=psy-ab&source=hp&pbx=1&oq=open&aq=0p&aqi=p-p2g2&aql=f&gs_sm=&gs_upl=&bav=on.2,or.r_gc.r_pw.,cf.o |
| 23:31.43 | SeRi | sb&fp=485b13d6987bbc3c&biw=1920&bih=964 |
| 23:33.27 | SeRi | http://pastebin.com/FXYadb0a |
| 23:34.16 | SeRi | https://forum.openwrt.org/viewtopic.php?id=566 |
| 23:34.58 | SeRi | http://luxik.cdi.cz/~devik/qos/htb/manual/userg.htm |
| 23:38.16 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-135-174.pools.arcor-ip.net) |
| 23:40.03 | dijib | http://pastebin.com/QbWQNTWy |
| 23:42.56 | dijib | http://pastebin.com/85FMPQmD |
| 23:49.21 | *** join/#asterisk DrkShadow (~andrew@host-72-175-240-62.static.bresnan.net) |
| 23:51.01 | DrkShadow | I have a sip trunk set up to grab one of our UK numbers.. it sends the number to the context "inbound-enabled" to make sure that that accountcode is allowed to receive calls. However, the account code _is_not_set_. |
| 23:51.02 | DrkShadow | http://pastebin.com/DStDdqvN |
| 23:51.29 | DrkShadow | What the hell? It's right there. I _know_ it's using that part of the sip.conf because if I change the context to simply "inbound" the call will come in to the "inbound" context. |
| 23:57.08 | DrkShadow | maybe it's not. In fact, it's not even going to the appropriate extension, despite being defined, in the inbound contect. At least, [inbound]<phoneno>,1,SET(CDR(accountcode)=windy) is never executed, yet it executes [inbound] <phoneno>,1,Goto(inbound-enabled,1) |
| 23:57.27 | DrkShadow | ... because the phone number was int here more than once.. |
| 23:58.33 | *** join/#asterisk Kuryoso (bd3d2321@gateway/web/freenode/ip.189.61.35.33) |
| 23:58.58 | DrkShadow | It's simply not listening to any trunk definition, and I don't know why. |
| 23:59.13 | Kuryoso | Hi people. Good night! Someone here from Brazil? |
| 23:59.29 | DrkShadow | kuryoso: "good night" = good bye |
