| 00:01.53 | *** join/#asterisk Tuxguy (n=jimi@65.184.197.243) |
| 00:01.58 | Tuxguy | does asterisk work w/ avaya? |
| 00:02.20 | Kobaz | Tuxguy: depends what you want to o |
| 00:02.22 | Kobaz | do |
| 00:02.51 | Kobaz | Tuxguy: if the avaya has t1, you can link up with t1, if it has sip, you can use sip, if it has analog, you can use analog... if it has cti, you can use cti |
| 00:03.09 | Tuxguy | oh |
| 00:04.20 | Kobaz | Tuxguy: were you expecting something more magical? |
| 00:04.26 | KyleK | KavanS: so PAP2 -> AsteriskA -> murphy? |
| 00:04.34 | Tuxguy | i dont even know what avaya is |
| 00:04.35 | Tuxguy | lol |
| 00:04.44 | Kobaz | it's a pbx |
| 00:05.10 | KavanS | KyleK, yep that's correct |
| 00:07.55 | KyleK | is there a user entry in sip.conf for the pap2 on murphy? the calls might come through as pap2@asteriska |
| 00:08.27 | *** join/#asterisk Dolfe (n=root@ool-457322ba.dyn.optonline.net) |
| 00:13.53 | *** join/#asterisk Octothorpe (i=octothor@pdpc/supporter/professional/octothorpe) |
| 00:14.57 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 00:15.19 | KavanS | KyleK, no there's not I figured the asterisk server would act like a proxy |
| 00:16.39 | KyleK | well it is, which is why calls are coming through as pap2@asteriska instead of pap2@pap2-ip-address |
| 00:19.36 | KavanS | yeah I just added it under sip.conf and no dice, same difference |
| 00:20.46 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 00:28.01 | *** join/#asterisk pbx1 (n=pbx1@203.82.38.122) |
| 00:30.09 | kn0x | Kobaz: problems fixed. only thing left is that Kernel panics on shutdown |
| 00:30.19 | kn0x | something to do with unloading zaptel/wanpipe |
| 00:30.44 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 00:35.38 | Kobaz | kn0x: yeah |
| 00:35.49 | Kobaz | kn0x: remove the dahdi script from your shutdown |
| 00:36.01 | Kobaz | kn0x: that's the easy fix |
| 00:36.10 | Kobaz | the problem is that dahdi is being unloaded before wanrouter |
| 00:36.15 | Kobaz | wanrouter needs to be unloaded first |
| 00:36.45 | Kobaz | the proper fix is to fix the order... but you dont really need to... modules dont need to be unloaded before shutdown |
| 00:40.32 | kn0x | oh |
| 00:40.49 | kn0x | how do i fix the router? |
| 00:40.51 | kn0x | i mean |
| 00:40.52 | kn0x | order |
| 00:40.53 | kn0x | lol |
| 00:41.45 | Kobaz | what's the output of: runlevel |
| 00:41.57 | kn0x | N 2 |
| 00:43.02 | Kobaz | edit /etc/rc2.d |
| 00:43.10 | Kobaz | look for files that start with K |
| 00:43.18 | Kobaz | move dahdi to be before wanrouter |
| 00:44.32 | kn0x | there are no K files in rc2.d |
| 00:44.40 | kn0x | there are some in the other runlevels |
| 00:45.19 | Kobaz | ah |
| 00:45.26 | Kobaz | look in /etc/rcS.d |
| 00:45.57 | Kobaz | are you running debian? |
| 00:45.59 | Kobaz | or ubuntu |
| 00:46.12 | kn0x | debian |
| 00:46.23 | kn0x | K29 wanrouter is in rc0 rc1 and rc6 |
| 00:46.43 | kn0x | ohh |
| 00:46.44 | Kobaz | okay so the kills are just in the shutdown runlevels |
| 00:46.52 | kn0x | oh |
| 00:46.55 | Kobaz | so in 0,1,6, fix the order |
| 00:46.57 | kn0x | K29wanrouter |
| 00:47.05 | kn0x | K30zaptel |
| 00:47.08 | Kobaz | yeap |
| 00:47.10 | kn0x | so it should be reversed? |
| 00:47.12 | Kobaz | that's bad |
| 00:47.15 | Kobaz | yeah reversed |
| 00:47.19 | Kobaz | oh wait |
| 00:47.20 | Kobaz | no |
| 00:47.22 | Kobaz | i'm on crack |
| 00:47.24 | Kobaz | no that's fine |
| 00:47.25 | kn0x | so, zaptel should shutdown first? |
| 00:47.29 | Kobaz | you want wanrouter stoping |
| 00:47.31 | Kobaz | and then zaptel |
| 00:47.38 | kn0x | oh. well its already like that. |
| 00:47.40 | Kobaz | okay so you don't have the problem that i think you have |
| 00:47.54 | Kobaz | okay just remove wanrouter zaptel from the shutdown |
| 00:49.43 | kn0x | okay rebooting |
| 00:52.03 | kn0x | Kobaz: okay. thanks. |
| 00:52.06 | kn0x | fixed it |
| 00:52.29 | kn0x | now if i could only fix this LVM shutdown failure mesg -_- |
| 00:53.48 | Kobaz | heh |
| 00:53.59 | Kobaz | now you're getting greedy |
| 00:54.09 | Kobaz | fixing all the errors, that's ridiculas! |
| 00:54.30 | KyleK | diminishing returns |
| 00:56.04 | kn0x | http://www.nabble.com/lvm-%2B-dm-crypt-%3D-shutdown-problem-(mount:---is-busy)-td15713092.html |
| 00:56.14 | kn0x | nahh lol im used to that im joking |
| 00:56.28 | kn0x | its something fucked up with debian ive noticed ever since i started using LVM |
| 00:56.39 | kn0x | its a chicken-and-the-egg scenario |
| 00:56.44 | Kobaz | i have a debian lvm setup, i dont think i have that problem |
| 00:56.44 | kn0x | ...sort of.. but backwards |
| 00:56.51 | Kobaz | is your root on lvm? |
| 00:57.04 | kn0x | Kobaz: yes |
| 00:57.11 | Kobaz | that's the problem |
| 00:57.14 | kn0x | only thing that isnt is /boot |
| 00:57.24 | kn0x | Kobaz: but thats the debian insatllers default |
| 00:57.49 | Kobaz | i have a seperate 20 gig partition (mirrored) that's root |
| 00:57.53 | Kobaz | and everything else is lvm |
| 00:58.27 | kn0x | yes. noted. good idea. |
| 00:59.40 | kn0x | first time you pick up the receiver on the fxs it doesnt pickup in asterisk |
| 00:59.44 | KyleK | yea I have a 16gb / and /home is lvm |
| 01:00.08 | kn0x | after a restart... |
| 01:00.22 | kn0x | everytime after that, once you pick up... there is dialtone |
| 01:00.38 | KyleK | so you gotta pick up a FXS or all of them once? |
| 01:01.10 | kn0x | i only have one fxs on the card |
| 01:01.22 | kn0x | and the very first time you pickup the phone after asterisk restarts |
| 01:01.25 | kn0x | there will be no dialtone |
| 01:01.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-1d04aae90a7a6ac2) |
| 01:01.42 | kn0x | subsequent pickups will go to dialtone without problem |
| 01:06.25 | *** join/#asterisk blkry (n=blkry@24-241-119-056.dhcp.gnvl.sc.charter.com) |
| 01:06.33 | KyleK | seems odd |
| 01:06.37 | kn0x | yes indeed |
| 01:07.38 | Kobaz | what asterisk version |
| 01:07.51 | Kobaz | there were some dialtone pickup problems fixed in like 1.4.22ish |
| 01:09.47 | kn0x | Kobaz: thats where im at |
| 01:09.51 | kn0x | asterisk 1.4.21 |
| 01:10.00 | kn0x | as per j00r advice :P |
| 01:10.20 | kn0x | actually theres this annoying noise in the background of the fxs |
| 01:11.21 | Kobaz | oh |
| 01:11.25 | Kobaz | i'm forgetting who is who |
| 01:11.32 | Kobaz | yeah |
| 01:11.37 | Kobaz | lemme look at the changelog |
| 01:11.49 | Kobaz | i'm doing some late night coding for a demo tomorrow |
| 01:11.50 | Kobaz | heh |
| 01:13.48 | Kobaz | well |
| 01:13.49 | Kobaz | first off |
| 01:13.53 | Kobaz | what line card is it? |
| 01:16.12 | kn0x | B600 |
| 01:16.20 | Kobaz | what company? |
| 01:16.23 | kn0x | sangoma |
| 01:16.26 | Kobaz | oh |
| 01:16.36 | Kobaz | b600? what's that a bri |
| 01:16.53 | Kobaz | oh yeah sangoma, we're working with wanpipe |
| 01:16.56 | Kobaz | heh, i'm tired |
| 01:17.30 | Kobaz | oh wow |
| 01:17.32 | Kobaz | that's new |
| 01:17.37 | Kobaz | little analog card |
| 01:17.47 | Kobaz | so |
| 01:17.51 | Kobaz | 1) is your phone crappy |
| 01:17.56 | Kobaz | 2) is your wiring crappy |
| 01:18.00 | Kobaz | 3) is your grounding crappy |
| 01:18.09 | kn0x | grounding on what? |
| 01:18.12 | Kobaz | the pc |
| 01:18.29 | kn0x | i dont have a special ground.. just the AC ground |
| 01:18.42 | kn0x | ill have to try with another phone and wires |
| 01:21.41 | Kobaz | that's the only think i can think of that would cause noise |
| 01:21.55 | Kobaz | unless you have a tv tuner or something in the same box, putting out noise into the pc case |
| 01:22.03 | Kobaz | is your cpu fan close to the card? |
| 01:23.32 | drmessano | IRQ |
| 01:23.52 | Kobaz | maybe |
| 01:23.58 | Kobaz | cat /proc/inturupts |
| 01:24.06 | Kobaz | /proc/interrupts |
| 01:34.12 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:47.16 | kn0x | Kobaz: what am i looking for in /proc/interrupts |
| 01:47.43 | kn0x | Kobaz: yes fan is close. it is a small form-factor machine |
| 01:53.14 | Kobaz | kn0x: you're looking for the analog card being on the same inturrupt as something else |
| 01:54.30 | kn0x | Kobaz: http://pastebin.ca/1411714 |
| 01:55.36 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:56.05 | Kobaz | 17: 2315799 0 IO-APIC-fasteoi wanpipe1 |
| 01:56.06 | Kobaz | looks good |
| 01:56.33 | Kobaz | do you have another phone to try? |
| 01:56.40 | kn0x | yeah |
| 01:56.46 | Kobaz | is it noisey? |
| 01:57.52 | *** part/#asterisk comprookie2000 (n=david@gentoo/contributor/comprookie2000) |
| 02:00.10 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:05.40 | kn0x | Kobaz: its better |
| 02:05.47 | kn0x | still hissing a bit |
| 02:05.51 | kn0x | like open air |
| 02:05.55 | kn0x | i guess thats normal |
| 02:06.05 | kn0x | but its a bit more than the co line |
| 02:07.29 | *** join/#asterisk tfrew (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 02:12.22 | Kobaz | it really comes down to the quality of power in the pc, and local inteferance |
| 02:12.36 | Kobaz | and quality of the line card, which... sangoma makes good stuff |
| 02:12.44 | Kobaz | but that's a new card |
| 02:13.07 | Kobaz | maybe it's got some circuits with inductance that shouldnt be there |
| 02:14.07 | kn0x | Kobaz: yeah. ill bring it up with them. |
| 02:14.10 | kn0x | they have good support |
| 02:14.18 | kn0x | thanks for the helkp 2day Kobaz |
| 02:14.18 | Kobaz | who knows... would need to hook up an occilascope |
| 02:14.21 | Kobaz | np |
| 02:17.07 | *** join/#asterisk hterag (n=chatzill@210.18.209.85) |
| 02:18.24 | hterag | G'day all... I am having an annoying issue no sip peers seem to be showing up but they are in users.conf.... and show up in the gui |
| 02:18.36 | Kobaz | users.conf is old... use sip.conf |
| 02:18.48 | Kobaz | see the example configs |
| 02:22.42 | hterag | hmm maybe its a asterisk-gui issue then |
| 02:23.15 | hterag | as all users/extensions show up in the gui but asterisk doesn't think they are there |
| 02:26.06 | Kobaz | yeah this chan isn't for asterisk-gui |
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| 02:40.05 | shyam_k | can i get smsq in 1.4.24? |
| 02:40.31 | shyam_k | its not there now.. should be there by default? or what should i do to get that? |
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| 03:38.10 | beernutz | hey, how can i simulate an incoming call from a command line client under linux? I have been messing with sipsak, but i cant seem to figure out the syntax. |
| 03:38.39 | beernutz | for some reason this is harder to google than i thought it would be |
| 03:39.00 | tfrew | beernutz: create a 7777 extension |
| 03:39.05 | tfrew | google the 7777 example |
| 03:39.17 | beernutz | coooool. thank you |
| 03:40.39 | tfrew | i take it your sarcastic |
| 03:40.50 | beernutz | sorry? |
| 03:41.08 | beernutz | i am googling it now |
| 03:41.24 | tfrew | don't mind me |
| 03:41.32 | tfrew | i'm pissed off at other random anon's |
| 03:41.39 | beernutz | ohhh lol |
| 03:43.05 | [TK]D-Fender | beernutz: "help originate" |
| 03:43.19 | beernutz | thank you Fender |
| 03:43.25 | beernutz | looking at that one now |
| 03:44.01 | [TK]D-Fender | beernutz: Of course you should simply go and install a softphone |
| 03:44.16 | beernutz | ya, i have that done too |
| 03:44.34 | beernutz | here is the deal: I am set up to talk to vitelity as my provider |
| 03:44.38 | beernutz | i register fine |
| 03:44.56 | beernutz | but when i try to make incoming callls using my cell phone to test the voip line |
| 03:45.15 | beernutz | it times out, and vitelity tell me it is getting 404 messages |
| 03:45.22 | beernutz | indicating a routing issue |
| 03:45.34 | beernutz | but the ODD thing is that it works sometimes. |
| 03:45.47 | beernutz | isnt it an all or nothing kind of thing? |
| 03:46.03 | [TK]D-Fender | beernutz: Means you should be enabling ISP DEBUG and watching the incoming call |
| 03:46.09 | [TK]D-Fender | SIP* |
| 03:46.10 | tfrew | traceroute your provider |
| 03:46.16 | beernutz | what i am attempting to do is diagnose it from the outside coming in |
| 03:46.23 | tfrew | *any* packet loss to them is a problem |
| 03:46.33 | beernutz | interesting point tfrew |
| 03:46.50 | [TK]D-Fender | PL is not the issue |
| 03:47.02 | [TK]D-Fender | beernutz: Go look at SIP DEBUG at * CLI for the calls |
| 03:47.10 | tfrew | your face is not the issue |
| 03:47.34 | *** part/#asterisk tfrew (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 03:47.35 | beernutz | thank you, ill look at that next |
| 03:48.10 | beernutz | may i ask however, si there a way to debug this from the outside? i want to see waht the conversation looks like from the outside coming in |
| 03:48.26 | [TK]D-Fender | beernutz: that makes no sense |
| 03:48.35 | beernutz | i supose you are right. lol |
| 03:48.45 | beernutz | i am trying to see what my provider sees |
| 03:48.50 | beernutz | or if i can replicate it |
| 03:49.15 | beernutz | from my side of the connection the logs are pretty bare, showing no connection attempts from them |
| 03:49.35 | beernutz | but i see successful registrations every few minutes |
| 03:49.43 | beernutz | from me to them |
| 03:50.12 | [TK]D-Fender | beernutz: This is what YOU see. they see what YOU answer. |
| 03:50.26 | [TK]D-Fender | beernutz: So you get to see the full conversation. There is not 3rd party here |
| 03:50.29 | [TK]D-Fender | no* |
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| 04:13.03 | beernutz | Fender: http://pastebin.com/mf726ab4 |
| 04:13.15 | beernutz | would you look at this with me please? |
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| 05:29.50 | Juggie | is there any dialplan func to check for the existance of a file? |
| 05:46.00 | *** part/#asterisk beernutz (i=beernutz@173-29-148-188.client.mchsi.com) |
| 05:47.15 | xorl | MaliutaLap: hey, so my phones work w/out using DHCP |
| 05:47.44 | xorl | MaliutaLap: moved the TFTP server in house to a crappy little temp machine in house until the new 1U arrives but the phones work again, just not using DHCP |
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| 06:47.57 | shyam_k | anyone tried with scribblej.com/svn 's astsphinx here? |
| 06:48.14 | shyam_k | its having provision for ngram models to recognize |
| 06:49.23 | shyam_k | i changed it to fsg the code is here swathanthran.in/astsphinx.c |
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| 07:32.10 | genin | mornin |
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| 07:53.17 | ruben23 | <PROTECTED> |
| 07:53.38 | ruben23 | is ip tunnneling a solution. |
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| 08:14.43 | Juggie | is it possible to set a variable inside a macro and then access it after the macro finishes? |
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| 08:17.36 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 08:34.38 | MaliutaLap | xorl: I am still willing to be something in the VLAN config is screwing your DHCP |
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| 08:47.36 | cyd777_wrk | hi |
| 08:49.04 | cyd777_wrk | i have a question: Can I use an external serial modem as inbound line device? I want to test somethings but first I don't want to by a fxo card |
| 08:50.50 | NirS | external modem? as in a device connected via a serial DB9 connector ? |
| 08:52.07 | NirS | hadn't seen a Serial modem in over 10 years |
| 08:52.29 | KyleK | if it supports Voice maybe |
| 08:52.56 | NirS | I don't think that's doable |
| 08:53.31 | NirS | it means that the voice needs to traverse between the modem and the machine via the serial link, and I'm not entirely sure if Zaptel/DAHDI have support for that |
| 08:53.47 | KyleK | it |
| 08:53.50 | KyleK | whoops |
| 08:53.54 | KyleK | it'd be a different driver |
| 08:53.56 | NirS | besides, as far as I know, the only MODEM based hardware supperted is a V.92 modem chip from Intel, |
| 08:54.21 | NirS | and I'm not familiar with External modems that utilize that specific shipset, however, as I never tried it - you got me there |
| 08:54.47 | NirS | here's a suggestion, if you really want to try out Asterisk, go on eBay and buy a single FXO card for a few dollars |
| 08:55.46 | KyleK | if it doesn't work out change the firmware and use it as a v92 modem? :) |
| 08:56.11 | KyleK | chan_modem |
| 08:57.18 | KyleK | cyd777_wrk: with a serial modem you might have only half duplex which means the modem has to switch which way audio is going |
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| 09:41.20 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 09:41.23 | foo | anyone use sipdroid on the g1? I keep getting timeout trying to connect to my asterisk box |
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| 09:43.30 | foo | If a sip client connects to my asterisk/trixbox system and times out... what do I need to adjust? I was told it might be something with nat. thanks |
| 09:50.34 | gr0mit | foo, sipdroid? |
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| 09:50.40 | gr0mit | where can i get it? |
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| 09:50.59 | foo | gr0mit: http://code.google.com/p/sipdroid/ |
| 09:51.13 | foo | gr0mit: going to play around with it now? let me know if you can get it to work |
| 09:53.15 | gr0mit | downloading... |
| 09:53.26 | foo | waits for you before he goes to sleep' |
| 09:54.00 | gr0mit | install unsuccessful :-( |
| 09:54.02 | pif | hi, anyone using OrderlyStats ? |
| 09:54.46 | foo | gr0mit: actually, it requires 1.5 |
| 09:55.03 | gr0mit | cupcake? |
| 09:55.04 | foo | gr0mit: you root your phone? |
| 09:55.06 | foo | gr0mit: yeash |
| 09:55.14 | gr0mit | nope phone is not rooted |
| 10:05.36 | tamiel | Hello, I have some congestion problems with Dahdi . I think the remote side is not releasing the line immediatly. I found many dahdi channels are stuck with PRI Flags: Resetting . |
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| 10:39.02 | ramonpeek | question: Does anyone know the reason why dialplan execution at the 'h' exten stops on <ZOMBIE> channels whilst it doesn't on normal hangups (Is this normal behaviour?) |
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| 10:57.32 | infernix | mnicholson: have you tried chan_mobile with android 1.5 yet? it worked fine with 1.1 but i'm getting only noise since the upgrade. probably due to the new bt stack in android |
| 10:57.48 | infernix | tries alignmentdetection=yes |
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| 11:10.26 | liri | is it possible to configure the directory of the meetme (conf rooms) recordings? I'd like to take it out of it's normal /var/lib/asterisk/sounds dir where all other sound files are present |
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| 11:44.35 | dpryo | Anybody using ExternalIVR()? |
| 11:44.51 | dpryo | Seems a bit too simple? |
| 11:45.41 | dpryo | If I want a call to be transfered when the caller enters a digit, I have to tell a channel-variable where I want to goto, and then do the rest in extensions? |
| 11:46.05 | dpryo | ExternalIVR() is basically a musicplayer? Plays music from playlists? :) |
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| 12:06.52 | angryuser | hello, when server is configured to accept unauthentificated calls, is there any option how the dtmf is handled for this calls (i want it set to rfc2833) ? |
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| 12:13.43 | orn | angryuser: Probably the [general] section of sip.conf |
| 12:22.43 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:22.52 | angryuser | orn, oh thx it was it ;) |
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| 13:03.51 | tamiel | When dahdi channel is flagged with "PRI Flags: Resetting", is it unavailable ? (here pastebin of a channel in this situation : http://pastebin.com/m48b229da ) |
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| 13:11.09 | EUS-Eric-DCaP | Sup guys |
| 13:11.33 | EUS-Eric-DCaP | Hey, is Jared Smith from Digium in here? |
| 13:14.14 | jaytee | EUS-Eric-DCaP, not at the moment but he does come in here occassionally |
| 13:14.31 | EUS-Eric-DCaP | Thanks, I was just wanting to say high. |
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| 13:25.22 | leifmadsen | EUS-Eric-DCaP: not in here, but you can find him in #asterisk-doc |
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| 13:27.28 | EUS-Eric-DCaP | hi, not high. |
| 13:27.30 | EUS-Eric-DCaP | haha |
| 13:27.32 | EUS-Eric-DCaP | didn't notice till now |
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| 14:07.53 | jaytee | "Sometimes, the only way to deal with a bully is to wind up and smack him upside the head with the Alvin and the Chipmunks Lunchbox of Justice." -Mentat |
| 14:08.06 | jaytee | now that's an excellent quote of the day |
| 14:08.27 | putnopvut | Ducktales lunchbox, more like. |
| 14:08.37 | jaytee | that would work too :-) |
| 14:08.39 | eppigy | hello |
| 14:08.42 | eppigy | i am dave |
| 14:08.44 | jaytee | hi dave! |
| 14:08.56 | eppigy | :D |
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| 14:17.01 | mintee | is there a list of causes somewhere? I'm getting "Channel 0/1, span 2 got hangup, cause 28" and can't find anything about cause 28 |
| 14:19.18 | file | they are isdn causecodes |
| 14:19.46 | file | http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
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| 14:19.55 | file | 28 is invalid number format |
| 14:20.04 | mintee | so i see |
| 14:21.24 | beek | morning jaytee |
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| 14:30.45 | jaytee | morning beek |
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| 14:35.32 | mintee | wow |
| 14:35.34 | mintee | http://www.bayhamsystems.com/asterisk.html |
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| 14:39.19 | jasonwoot | is there a quick way to pause all queue members after a reboot? |
| 14:40.15 | eppigy | use realtime config |
| 14:40.24 | eppigy | and set all agents in the table to paused |
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| 14:57.10 | jaytee | morning brian |
| 14:57.50 | telnettech | I have a question.......if i have 4 and 5 digit extension numbers, does the pattern match have to be 5 digits or do i need to build both 4digit and 5digit pattern matches even though they are doing the same thing in dialplan? |
| 14:57.56 | telnettech | morning jaytee |
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| 14:59.53 | [TK]D-Fender | telnettech: ther is no "4 or 5" pattern possible. though there is a "4 or LONGER" options |
| 14:59.53 | Nugget | telnet is eeeeeeevil! |
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| 15:00.45 | telnettech | so if i have my pattern match as _7xxx that will take care of both 4 and 5 digit extensions? |
| 15:03.00 | [TK]D-Fender | telnettech: Clearly NO. |
| 15:03.37 | telnettech | so i need a 5digit like _7xxxx for 5 digit extensions |
| 15:04.02 | [TK]D-Fender | telnettech: those "x" do mean something you know... You really need to go read chapter 5 again... |
| 15:05.09 | telnettech | the "x" mean any digits from 0 thru 9.. |
| 15:06.06 | telnettech | so in the pattern match _7xxxx it means that the number will start with "7" and 4 more digits after it with each digit being from 0 thru 9 |
| 15:06.52 | telnettech | but in the pattern match _7xxx it means that it starts with 7 with 3 more digits afterwards from 0 thru 9 |
| 15:07.41 | telnettech | so I would assume ( ass u me....haha!!!) that I need to include a pattern match for both sets of extensions--4 digit and 5 digit extensions |
| 15:10.07 | [TK]D-Fender | telnettech: Yes, its every bit as cut & dry as it sounds. 2 Pattern matchs required |
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| 15:10.30 | telnettech | ok...Thanks TK |
| 15:10.37 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 15:11.26 | Pagautas | or maybe use _7. :) |
| 15:12.47 | Pagautas | is there anyone working with quintum hardware? |
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| 15:13.27 | ruben23 | <PROTECTED> |
| 15:14.53 | EUS-Eric-DCaP | ok |
| 15:14.55 | EUS-Eric-DCaP | what is your firewall? |
| 15:19.03 | ruben23 | EUS-Eric-DCaP: i got a centos box |
| 15:19.08 | ruben23 | iptables |
| 15:19.12 | EUS-Eric-DCaP | oh. |
| 15:19.36 | EUS-Eric-DCaP | so you need to port forward tcp 5060 and udp 10000-20000 to your pbx from your external IP |
| 15:19.58 | EUS-Eric-DCaP | in your SIP.conf general file, you need to define "externip=" |
| 15:20.13 | [TK]D-Fender | ~sipnat |
| 15:20.14 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:20.17 | EUS-Eric-DCaP | for your PBX, and then in the sip peer entry you need to include nat=yes |
| 15:20.17 | [TK]D-Fender | ^^^^^ |
| 15:20.28 | ruben23 | what would be the value externip..? |
| 15:20.47 | EUS-Eric-DCaP | the external IP that the PBX will take registrations with. |
| 15:21.00 | EUS-Eric-DCaP | fender, do you have a list of all those ~ commands? |
| 15:21.05 | EUS-Eric-DCaP | I'd like to look over them |
| 15:23.09 | ruben23 | EUS-Eric-DCaP: but my asterisk is in private ip. |
| 15:23.39 | EUS-Eric-DCaP | that's why your port forwarding from an external IP through your firewall to your asterisk box |
| 15:24.03 | EUS-Eric-DCaP | externip is telling asterisk what it's external IP is so it knows how to NAT and send RTP |
| 15:25.34 | ruben23 | ok |
| 15:28.38 | [TK]D-Fender | EUS-Eric-DCaP: What would you be looking for? |
| 15:28.48 | EUS-Eric-DCaP | the ~sipnat |
| 15:28.58 | EUS-Eric-DCaP | commands, that make infobot spit out an article. |
| 15:29.02 | [TK]D-Fender | EUS-Eric-DCaP: You can alrady see it... iys right there :p |
| 15:29.08 | EUS-Eric-DCaP | hahah. |
| 15:29.11 | EUS-Eric-DCaP | no, but all of them |
| 15:29.15 | EUS-Eric-DCaP | I assume there's more, right? |
| 15:29.18 | EUS-Eric-DCaP | ~sipnat |
| 15:29.19 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:29.26 | [TK]D-Fender | EUS-Eric-DCaP: I maintain them, and yes I keep the list |
| 15:29.36 | EUS-Eric-DCaP | can I see em? |
| 15:29.50 | EUS-Eric-DCaP | or is that you maintain dominance over the IRC channel? |
| 15:29.52 | [TK]D-Fender | EUS-Eric-DCaP: http://www.aocomputing.net/jbot.txt |
| 15:29.53 | EUS-Eric-DCaP | hhahah |
| 15:30.05 | [TK]D-Fender | EUS-Eric-DCaP: that too :) |
| 15:30.21 | [TK]D-Fender | EUS-Eric-DCaP: Slightly outdated copy. |
| 15:30.28 | [TK]D-Fender | EUS-Eric-DCaP: but covers most of them |
| 15:31.03 | [TK]D-Fender | EUS-Eric-DCaP: these are the ones I maintain at least,t here are tons of others |
| 15:31.41 | EUS-Eric-DCaP | !blf |
| 15:31.45 | EUS-Eric-DCaP | ~siphints |
| 15:31.58 | EUS-Eric-DCaP | I'm curious to read more about how BLFs and HINTS actually work |
| 15:32.21 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:35.31 | [TK]D-Fender | ~blf |
| 15:35.32 | infobot | from memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 15:35.42 | [TK]D-Fender | EUS-Eric-DCaP: And there is a decent page on the WIKi for this |
| 15:35.58 | EUS-Eric-DCaP | voip-info? |
| 15:36.06 | [TK]D-Fender | yes |
| 15:36.07 | EUS-Eric-DCaP | I don't need to know how to set them up, I know that |
| 15:36.11 | EUS-Eric-DCaP | but how they actually function. |
| 15:36.20 | EUS-Eric-DCaP | like the SIP messages they pass back and forth. |
| 15:36.32 | [TK]D-Fender | EUS-Eric-DCaP: Maybe you could be a little less generic. Perhaps even a LOT. |
| 15:36.45 | [TK]D-Fender | EUS-Eric-DCaP: Look at your SIP debug. |
| 15:36.52 | [TK]D-Fender | EUS-Eric-DCaP: It isn't hidden you know. |
| 15:37.13 | EUS-Eric-DCaP | I've done that, and I see what its doing, but I wanted to read some kind of man page that discusses in detail what's going on. |
| 15:37.38 | EUS-Eric-DCaP | I'm having an interesting problem at a client with a very basic network/pbx and I can't quite figure out what it is. |
| 15:37.42 | [TK]D-Fender | EUS-Eric-DCaP: You are still entirely too generic. |
| 15:37.47 | EUS-Eric-DCaP | Ok. |
| 15:38.01 | EUS-Eric-DCaP | When a call comes in, and a sip hint changes to RINGING, |
| 15:38.12 | EUS-Eric-DCaP | what is the PBX supposed to send out to the peers. |
| 15:38.16 | EUS-Eric-DCaP | And how do they respond. |
| 15:38.36 | [TK]D-Fender | EUS-Eric-DCaP: there is no RESPONSE. Its a NOTIFICATION |
| 15:38.36 | EUS-Eric-DCaP | When the state changes to INUSE or back to idle what does the PBX send out. |
| 15:38.56 | [TK]D-Fender | EUS-Eric-DCaP: And again this is something you could ahve answered for yourself in the span it took to ask, |
| 15:39.01 | EUS-Eric-DCaP | well, I'd like to learn exactly what both end points are doing. |
| 15:39.05 | file | SIP NOTIFY messages with a content type that match what the subscription asked for |
| 15:39.18 | file | the content describes the state. |
| 15:39.20 | [TK]D-Fender | eurSetup a hint on an device. place call, watch debug, hangup, watch debug |
| 15:39.42 | Dave-Network-plu | NEIN |
| 15:39.53 | EUS-Eric-DCaP | Thanks for that, but like I said, I'm having a problem with it, so obviously it's not working right. |
| 15:39.56 | [TK]D-Fender | Dave-Network-plu: WAHT ARE YOU DOING DAVE? |
| 15:39.58 | [TK]D-Fender | :p |
| 15:40.00 | EUS-Eric-DCaP | so doing a sip debug does me no good. |
| 15:40.13 | Dave-Network-plu | I want my network plus cert in my nick |
| 15:40.19 | [TK]D-Fender | EUS-Eric-DCaP: Why not? yous ee the complete message in there |
| 15:40.19 | Dave-Network-plu | so i can look totally awesome |
| 15:40.20 | EUS-Eric-DCaP | and I'd be looking at a huge mess of sip data. |
| 15:40.38 | [TK]D-Fender | EUS-Eric-DCaP: you can restrict by IP you know.. |
| 15:40.43 | Dave-Comptia | YES |
| 15:40.44 | EUS-Eric-DCaP | Yes, I know that. |
| 15:40.59 | [TK]D-Fender | EUS-Eric-DCaP: This is a 5 second test. |
| 15:41.10 | [TK]D-Fender | EUS-Eric-DCaP: There really is no excuse for this. |
| 15:41.19 | Dave-Comptia | What is your dcap number |
| 15:41.26 | Dave-Comptia | I am having your shit revoked |
| 15:41.53 | file | language :P |
| 15:41.58 | file | tickles Katty |
| 15:42.00 | Dave-Comptia | apologies sir |
| 15:42.11 | [TK]D-Fender | EUS-Eric-DCaP: How is it you can't enable sip debug on a single device, place a call and look at what notifications it was sent? |
| 15:42.16 | EUS-Eric-DCaP | Jesus. |
| 15:42.17 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:42.18 | EUS-Eric-DCaP | Of course I can do that. |
| 15:42.26 | [TK]D-Fender | EUS-Eric-DCaP: So whats the problem? |
| 15:42.38 | EUS-Eric-DCaP | I just want to look at some document that explains how they work. |
| 15:42.49 | file | that is complicated |
| 15:42.51 | EUS-Eric-DCaP | I can't find any documentation like that. |
| 15:43.01 | Dave-Comptia | http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%3A%2F%2Fwww.ietf.org%2Frfc%2Frfc3261.txt&ei=_gz_SaPPOsqDtgeS9OmiDQ&usg=AFQjCNGUbUGCyLDYylujSyF34xErxBapug |
| 15:43.05 | file | because there are different content types that can be used |
| 15:43.14 | Dave-Comptia | RFC 3261: SIP: Session Initiation Protocol |
| 15:43.16 | [TK]D-Fender | EUS-Eric-DCaP: Work HOW? Where * maintains a state? what the state #'s it passes on are? |
| 15:43.18 | file | and devices implement different ones |
| 15:43.24 | [TK]D-Fender | EUS-Eric-DCaP: And yes, this is in the RFC |
| 15:43.25 | file | at least at the protocol level. |
| 15:43.31 | EUS-Eric-DCaP | Ok, so I have some aastra phones. |
| 15:43.41 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:43.52 | EUS-Eric-DCaP | Somehow, the messages telling them a BLF is back to idle from a Ringing state isn't updating |
| 15:44.16 | EUS-Eric-DCaP | I see the sip messages going out from the PBX, but I'm not sure they the phone isn't updating the BLF |
| 15:44.24 | EUS-Eric-DCaP | That's it. |
| 15:44.38 | [TK]D-Fender | EUS-Eric-DCaP: This sounds like your phone has a problem, not *. |
| 15:44.56 | EUS-Eric-DCaP | Likely. |
| 15:45.12 | [TK]D-Fender | EUS-Eric-DCaP: And thats like triple or quadruple negative statement there... |
| 15:45.25 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:45.26 | EUS-Eric-DCaP | What? |
| 15:45.27 | [TK]D-Fender | EUS-Eric-DCaP: Does * send out the state change packet to it or not? |
| 15:45.44 | eppigy | i like to be positive |
| 15:47.28 | [TK]D-Fender | eppigy: A proton walks into a convention hall full of electrons and says "Uh ho, I think I'm in the wrong room". An eletron turns to him and says "Are you sure?". Proton responds "I'm positive!" |
| 15:50.13 | jdblack | please tell me you don't tell that joke in public |
| 15:50.40 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:50.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:51.54 | eppigy | [TK]D-Fender: LOL |
| 15:51.56 | KavanS | lol |
| 15:51.58 | KavanS | that's a good joke |
| 15:52.21 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:53.00 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
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| 16:05.22 | jad_jay | Hi all |
| 16:06.21 | jad_jay | I have a problem with the voices messages, i can't hear them from the beginning i just can hear till the middle of the sentence... |
| 16:07.11 | jad_jay | I put a wait of 10 seconde before but it is the same, i have to put two "hello world" before to be sure to heard something |
| 16:07.51 | EUS-Eric-DCaP | So I have a SIP debug from the PBX and a test phone |
| 16:09.30 | EUS-Eric-DCaP | so now, what do I look for? |
| 16:09.49 | EUS-Eric-DCaP | I guess no one has any information about how Aastra phones and Asterisk work together for hints, right? |
| 16:09.49 | [TK]D-Fender | EUS-Eric-DCaP: the NOTIFY's that * passes during call progress |
| 16:10.07 | [TK]D-Fender | EUS-Eric-DCaP: Same as an other SIP device |
| 16:10.24 | jad_jay | Maybe i could try with the different silence sounds |
| 16:10.31 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-213-245.lns10.mel6.internode.on.net) |
| 16:11.10 | [TK]D-Fender | jad_jay: Smoe clients are slower at setting up RTP streams for instance which can cause the start of calls of that type to seem cut-off. |
| 16:11.20 | [TK]D-Fender | jad_jay: Usually its only a second or two |
| 16:19.15 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:21.02 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 16:21.30 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:21.35 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:21.59 | *** join/#asterisk jtodd (n=jtodd@dslb-088-072-212-085.pools.arcor-ip.net) |
| 16:21.59 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:23.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:25.39 | Juggie | anyone have a suggestion on the best way to check via dialplan if a file exists... |
| 16:25.57 | [TK]D-Fender | Juggie: "sore show functions like STAT" |
| 16:26.02 | Juggie | i'd like to do it in a gosub/macro and i can use shell() but i'm tied to 1.4 |
| 16:26.54 | Juggie | [TK]D-Fender, what app/function provides that my install (trunk 1.6) doesnt have it. |
| 16:27.18 | Juggie | nm caps would help :) |
| 16:28.03 | [TK]D-Fender | Juggie: Yes... they would |
| 16:28.12 | Juggie | well thats handy :) |
| 16:30.17 | *** join/#asterisk Chuggs (n=tadd@s142-179-186-158.ab.hsia.telus.net) |
| 16:30.58 | jad_jay | [TK]D-Fender: Thanks I tried the silences sounds but it doesn't works, maybe with something like a "beeep" will did the trick |
| 16:31.10 | jad_jay | [TK]D-Fender: Or a song... |
| 16:32.11 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 16:34.34 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-735c9de45b202b28) |
| 16:35.00 | *** part/#asterisk awk_r (n=awk_r@nat/digium/x-735c9de45b202b28) |
| 16:35.02 | *** join/#asterisk Ziaeon (n=acabrera@75-149-177-2-Miami.hfc.comcastbusiness.net) |
| 16:35.04 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-735c9de45b202b28) |
| 16:36.31 | Ziaeon | Does asterisk 1.4 automatically use multicore cpu's? I have one asterisk box that takes about 20 concurrent calls, all of which are recorded, and sometimes cpu peaks and the quality suffers. It's a 64bit Core 2 Xeon 3.3ghz and 2gigs of ram. I recompiled the kernel to make sure it uses Core 2/ Xeon but asterisk still seems to chug on one core instead of utilizing all 4. |
| 16:36.54 | Ziaeon | Asterisk will often report up to 50 percent usage in TOP where my actual CPU usage is only about 7% |
| 16:37.45 | Qwell | Ziaeon: the kernel handles all of that. |
| 16:37.49 | *** join/#asterisk crevetor (n=crevetor@bureau.ubity.com) |
| 16:38.03 | Qwell | Asterisk is a multi-threaded application though, yes. |
| 16:38.44 | crevetor | Does anybody know how I could see what is causing latency in the treatment of sip packet by asterisk ? |
| 16:38.57 | Ziaeon | So can I assume theres something missing in my kernel conf if asterisk reports X amount of usage and my overall CPU usage is lower? IE: It's not properly taking advantage of all 4 cores? Or is this mostly the way it looks and I'm just asking too much of this one box? |
| 16:39.04 | Qwell | crevetor: standard network analysis tools |
| 16:39.34 | crevetor | Qwell: it seems to be "inside asterisk" |
| 16:40.01 | Qwell | "seems to be" or "is"? |
| 16:40.05 | Qwell | verify. |
| 16:40.23 | crevetor | Ok I'll check |
| 16:43.21 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:45.02 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 16:45.52 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:47.28 | mnicholson | infernix, no, haven't heard any reports of testing with 1.5 yet |
| 16:47.32 | *** join/#asterisk Dovid (n=annon@ool-4355e297.dyn.optonline.net) |
| 16:48.01 | Dovid | TK: What version of asterisk r u working with now ? |
| 16:49.01 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 16:49.39 | Dovid | Anyone from "Digium Land" here ? |
| 16:50.15 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 16:51.08 | tfrew | is that like disney land? |
| 16:52.22 | Dovid | lol |
| 16:52.28 | Dovid | there is an error on the site. |
| 16:54.04 | [TK]D-Fender | Dovid: I'm on 1.2 at work which I can't get around to upgrading (complicated), and 1.6.0.X at home |
| 16:54.27 | Dovid | ok. i have a test box. was wondering about 1.6.1.0 |
| 16:59.38 | tfrew | go with 1.6.x |
| 16:59.56 | eppigy | I hope you are doing well. |
| 17:03.27 | *** join/#asterisk nealix (n=np20433@nat/sun/x-d2b4543db217d42e) |
| 17:03.53 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 17:09.26 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-7626339ae1b43f6d) |
| 17:24.18 | *** join/#asterisk empiric (n=empiric@116.71.60.89) |
| 17:28.41 | Dovid | What does Digium have against blonde's ? |
| 17:28.42 | Dovid | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.0 |
| 17:28.55 | Dovid | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 |
| 17:28.55 | Dovid | <PROTECTED> |
| 17:28.55 | Dovid | <PROTECTED> |
| 17:31.38 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 17:32.38 | eppigy | Dovid: I think that is a typo |
| 17:32.54 | eppigy | they clearly discriminate against the blind |
| 17:34.19 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 17:34.33 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 17:49.26 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
| 17:49.49 | cjk | hi, is there a variable i can use in the dialplan that shows me which leg of my call is responsible for the hangup? |
| 17:50.49 | infernix | mnicholson: i'll keep you posted with my findings, will test some more after dinner |
| 17:51.51 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 17:52.21 | mnicholson | infernix, ok |
| 17:54.47 | Dovid | what version of spandsp does 1.6.X require ? |
| 17:54.51 | *** join/#asterisk pnlarsson (n=nick@c83-249-198-63.bredband.comhem.se) |
| 17:55.18 | Dovid | for app_fax |
| 17:55.52 | pnlarsson | What could be the reason for asterisk to hang when loading chan_iax2.so? Using both 1.2 and 1.4.svn |
| 17:56.00 | jasonwoot | with UnPauseQueueMember, can I specify an extension lieu of and agent ID at "interface" |
| 17:56.23 | pnlarsson | Can i make asterisk to drop core to see what the issue is? |
| 17:56.37 | pigpen | Hi all. I moved to asterisk 1.6.0.6 with dahdi 2.1.0.4 a few months ago. Using the TDM4xx card, no hw echo cancelation. I use this system at my house (6 polycom phones) for testing. |
| 17:56.47 | pigpen | I figure if the wife doesn't kill me, it must be running ok. |
| 17:56.59 | pigpen | Well, I am having issues with Echo Cancelation: chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 4 (No such device) |
| 17:57.28 | pigpen | I have Asterisk running in many locations from 5 users to 500, but this is my first move to 1.6 |
| 17:58.47 | pigpen | My dahdi system.conf references 1 - 3 as fxoks, and 4 as fxsks. |
| 18:00.22 | nealix | Newbie getting started: For my first IP phone set to learn, do you guys prefer Cisco 7960 or Polycom Ip-430 ? |
| 18:00.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:00.39 | pigpen | nealix, polycom |
| 18:01.05 | Chainsaw | leifmadsen: Can you please get your devs to stop closing unfixed bugs? http://bugs.digium.com/view.php?id=14671 |
| 18:01.12 | pigpen | I have echocanceller=mg2,4 |
| 18:01.32 | nealix | Is the polycom simply easier to set up, or superior in sound quality and features? |
| 18:01.43 | pigpen | with running dahdi_cfg -c system.conf -d 9 -f -x: DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22) |
| 18:01.55 | Chainsaw | leifmadsen: All that has to be done is applying this one patch. Practically a one-liner. I've signed the agreement. Do I really have to take out a support contract to get you guys to apply a diff? |
| 18:02.09 | pigpen | nealix, great sound, eaiser to deploy, and they run sip well. |
| 18:02.20 | pigpen | Cisco sees SIP as a bastard codec. |
| 18:02.20 | nealix | thanks |
| 18:02.35 | Chainsaw | pigpen: (SIP isn't a codec) |
| 18:02.50 | nealix | protocol, whatever :-) |
| 18:03.09 | pigpen | Chainsaw, bla, bla..sorry, not enough caffeine yet. |
| 18:03.13 | nealix | I'll hunt for the Polycom, that makes sense |
| 18:04.00 | pigpen | The 430's are ok, but I got very used to the 601/650's |
| 18:04.26 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:04.43 | pigpen | A great headset for them is the Jabra 9350 with the special cable you can get for it (I forget what it is called, and I KNOW Chainsaw will call me on it) |
| 18:05.19 | leifmadsen | Corydon76-dig: see above msg from Chainsaw |
| 18:05.51 | cjk | how can i debug hangups and check who is responsible for the hangup for calls from iax to zap |
| 18:06.10 | Chainsaw | Corydon76-dig: I am the Gentoo package maintainer. The other report is similar. If you blame my ebuild, be specific. |
| 18:06.42 | pigpen | Chainsaw, Gentoo Package Maintainer?? |
| 18:06.52 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:07.02 | pigpen | Chainsaw, know pfeifer@gentoo.org? |
| 18:07.17 | Chainsaw | pigpen: I don't think pfeifer's a dev anymore. |
| 18:07.35 | Chainsaw | pigpen: Of Asterisk, yes. |
| 18:07.42 | pigpen | correct, he isn't active in the sense. But I know he is still involved. |
| 18:07.46 | pigpen | He is my business partner. |
| 18:08.04 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:08.07 | pigpen | And yes, we only run Gentoo. |
| 18:08.57 | pigpen | This is good to know. We have had to patch asterisk numerous time in the past. It would be nice to get the patches to the right person. |
| 18:10.18 | jaytee | <PROTECTED> |
| 18:10.26 | Chainsaw | pigpen: I'm trying but I can't claim great success. |
| 18:10.30 | *** join/#asterisk grantm (n=grantm@174.46.115.135) |
| 18:10.37 | pigpen | heh, moving target eh? |
| 18:10.46 | foo | If a sip client connects to my asterisk/trixbox system and times out... what do I need to adjust? I was told it might be something with nat. thanks |
| 18:10.59 | Chainsaw | jaytee: That's because their RPM applies all the patches and they never send them upstream. |
| 18:11.08 | jaytee | qualify=yes |
| 18:11.13 | Chainsaw | jaytee: Just because you don't see it doesn't mean it doesn't happen :) |
| 18:11.16 | pigpen | jaytee, we have some rather large installations, that run across some odd issues. |
| 18:11.59 | pigpen | Chainsaw, any idea why my echocancel isn't working?? |
| 18:12.13 | Chainsaw | pigpen: Without seeing your config? No. |
| 18:12.22 | pigpen | before 1.6 echo cancel just worked. |
| 18:12.42 | pigpen | I can take care of that. |
| 18:13.17 | Chainsaw | pigpen: Disclaimer though, my TDM410 doesn't have the echo cancellation module installed. All I do over analog lines is accept inbound fax. |
| 18:13.38 | KyleK | mmmmm faxxx |
| 18:13.39 | jaytee | pigpen, are you using hardware echo cancellation or software? |
| 18:13.46 | pigpen | no hardware. |
| 18:13.56 | pigpen | I have had bad luck with the. |
| 18:13.59 | pigpen | s/the/them |
| 18:15.24 | jaytee | in 1.6 the echo cancel software modules are dynamic. You don't need to recompile dahdi to change software echo cancellers but it needs to be in your system.conf file i.e. echocanceller=mg2,1-4 where 1-4 are the channels as an example |
| 18:16.10 | Chainsaw | KyleK: Fax can be so annoying though. |
| 18:16.25 | pigpen | http://pastebin.com/m7f096f7 |
| 18:16.37 | pigpen | Chainsaw, we are using iaxmodem with hylafax. |
| 18:16.45 | pigpen | I have been using this for about 3 years. |
| 18:16.45 | Chainsaw | KyleK: Brother fax (MFC-9840CDW) -> Patton SmartNode 4118 -> Patton SmartNode 4634 -> BT ISDN BRI -> Digium TDM400 |
| 18:16.46 | jaytee | I saw the new announcement about Digium's fax addon with 1 free license. looked interesting but I'm running 64 bit :-( |
| 18:17.08 | Chainsaw | KyleK: If my outbound call follows that chain, the two fax machines never negotiate succesfully. I hear CNG, but never CED. |
| 18:17.14 | pigpen | Chainsaw, ok, see the above pastbin, pretty basic and generic. |
| 18:18.01 | pigpen | makes me wonder if the ver of dahdi is borked. |
| 18:18.22 | Chainsaw | jaytee: I think we're all running 64-bit by now. I certainly am. |
| 18:19.19 | jaytee | yeah, but neither Digium's new fax piece is supported now and Lumenvox's speech engine didn't when I first started developing our speech enabled IVR |
| 18:19.32 | jaytee | only on 32bit |
| 18:20.03 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 18:20.15 | pigpen | yeah, no dahdi updates. |
| 18:23.51 | *** join/#asterisk Corydon76-lap (n=Corydon7@pdpc/supporter/bronze/Corydon76-home) |
| 18:23.51 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 18:30.13 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 18:32.14 | Corydon76-lap | Chainsaw: in the future, please file reports only with our plain-jane source, NOT packager builds and NO distro-supplied patches. |
| 18:32.46 | Chainsaw | Corydon76-dig: So you don't want me to try to submit our patching upstream. Duly noted. |
| 18:32.56 | *** part/#asterisk pnlarsson (n=nick@c83-249-198-63.bredband.comhem.se) |
| 18:33.18 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:33.24 | Corydon76-lap | We cannot read your mind and know that your extras have nothing to do with a problem. In many cases, the patches are the problems, themselves, and we'd rather not be running down problems that aren't in our code |
| 18:35.27 | Chainsaw | Corydon76-dig: It'd be awesome if you could post such things to the bug tracker next time. |
| 18:35.37 | Chainsaw | Corydon76-dig: Because all I saw was a bug gathering bitrot. |
| 18:36.00 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:36.01 | Corydon76-lap | Chainsaw: it's in our reporting guidelines |
| 18:36.30 | Corydon76-lap | And in 95% of cases, telling people to consult with the packager is the right thing to do |
| 18:36.36 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:37.02 | Chainsaw | Corydon76-dig: Sure. I don't mind. Just tell me of that when you close it. |
| 18:37.06 | Corydon76-lap | I'll make a note for next time that you're the Gentoo maintainer |
| 18:38.14 | *** join/#asterisk ntbourey (n=ntbourey@c-76-110-3-120.hsd1.fl.comcast.net) |
| 18:38.19 | ntbourey | Hello everyone |
| 18:38.49 | ntbourey | I was wondering if someone might assist me with a problem I am having |
| 18:39.13 | Chainsaw | We can try. You're going to have to tell us what the problem is though. |
| 18:39.19 | ntbourey | Sure thing |
| 18:39.20 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:39.20 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:39.35 | ntbourey | I have asterisk set up with a bunch of SIP users |
| 18:39.37 | [TK]D-Fender | Chainsaw: Next you might even expect that description to be detailed. How droll |
| 18:39.47 | ntbourey | and softphones for each account |
| 18:39.53 | pigpen | Chainsaw, are the echo cancelation modules a kernel option (running 2.6.26-hardened-r9) ? |
| 18:39.55 | ntbourey | I can dial another sip address |
| 18:40.10 | ntbourey | but when the user accepts the call no audio gets transfered |
| 18:40.27 | Chainsaw | pigpen: No, they'd be a DAHDI option. |
| 18:40.33 | [TK]D-Fender | ntbourey: Sounds like a networking (NAT / reinvite) issue. |
| 18:40.41 | Chainsaw | pigpen: You'd compile them as kernel modules, but they're an external module, built through the dahdi ebuild. |
| 18:40.48 | [TK]D-Fender | ntbourey: firewall possibly. Describe the networking between them |
| 18:41.02 | ntbourey | We are all connected to a router |
| 18:41.09 | ntbourey | generally with DHCP |
| 18:41.11 | pigpen | Chainsaw, k, I am just getting nowhere with this. |
| 18:42.23 | Chainsaw | pigpen: There are USE-flags to select the echo cancellers. |
| 18:42.30 | Chainsaw | pigpen: Try and emerge -pv dahdi and see what's on. |
| 18:42.37 | pigpen | hmm.yeah, that could be it. |
| 18:42.51 | pigpen | before we patched it for mg2 |
| 18:43.39 | pigpen | Heh...doesn't say a word....gee, that could be it. |
| 18:43.55 | ntbourey | Any thoughts? |
| 18:44.07 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 18:44.41 | pigpen | ntbourey, give it a few min, many of us are involved.... |
| 18:44.52 | ntbourey | Okay |
| 18:46.04 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:46.21 | [TK]D-Fender | ntbourey: All all parts of these calls WITHIN that local LAN? |
| 18:46.29 | ntbourey | Yes |
| 18:46.31 | *** part/#asterisk Corydon76-lap (n=Corydon7@pdpc/supporter/bronze/Corydon76-home) |
| 18:46.46 | ntbourey | Nothing is going out to an external sip provider |
| 18:46.57 | [TK]D-Fender | ntbourey: then THEY are not part of your local LAN |
| 18:47.06 | [TK]D-Fender | ntbourey: I asked if ALL ends were |
| 18:47.14 | [TK]D-Fender | ntbourey: Go read the guide : |
| 18:47.17 | [TK]D-Fender | ~sipnat |
| 18:47.18 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:47.19 | [TK]D-Fender | ^^^^^^ |
| 18:47.43 | ntbourey | And I said yes they are all within that local lan |
| 18:49.32 | ntbourey | Yeah that only helps if I am trying to call outside of my network |
| 18:49.33 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 18:49.36 | [TK]D-Fender | ntbourey: This external SIP provider (ITSP) doesn't seem to be |
| 18:49.44 | ntbourey | I dont have one |
| 18:49.58 | ntbourey | ntbourey: Nothing is going out to an external sip provider 2:47PM |
| 18:50.10 | [TK]D-Fender | ntbourey: Sorry, I seem to have misread. Check the firewall on your server |
| 18:50.18 | ntbourey | There is none |
| 18:50.34 | [TK]D-Fender | ntbourey: then next those on each mhacnine |
| 18:50.54 | [TK]D-Fender | ntbourey: Foloowing that check your sound card and any audio level monitoring in your softphone apps |
| 18:51.15 | ntbourey | All of the clients are macs |
| 18:51.26 | ntbourey | all are properly set up |
| 18:51.35 | ntbourey | I am watching a video on my headset and can hear it |
| 18:51.43 | ntbourey | and I can see that I its also recording |
| 18:51.51 | pigpen | Chainsaw, I thought the package would say the use flag, but I am not finding it, would you mind hinting to me a bit...? |
| 18:53.47 | *** join/#asterisk joseph__ (i=CK@89.249.221.16) |
| 18:53.58 | joseph__ | gi all |
| 18:54.41 | pigpen | gi |
| 18:55.10 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:55.12 | joseph__ | how to allow early media like voicemail if i used dial(sip/${EXTEN}@TRUNK,30,m) |
| 18:55.40 | Meaw | do you guys know a good page about how to install asterisk + sangoma A102 |
| 18:56.24 | joseph__ | my scenario is userA -->*--->termination GW |
| 18:56.55 | joseph__ | user will hear an MOH when dialing |
| 18:58.53 | foo | If a sip client connects to my asterisk/trixbox system and times out... what do I need to adjust? I was told it might be something with nat. thanks |
| 18:59.19 | joseph__ | pigpen any idea ? |
| 18:59.53 | ntbourey | foo: you might need to set your sip proxy to your asterisk server |
| 19:00.41 | [TK]D-Fender | foo: [15:00]<[TK]D-Fender>?? nat |
| 19:00.43 | [TK]D-Fender | [15:00]<coopbot>nat: If behind a NAT, create an /etc/asterisk/sip_nat.conf file with AT LEAST these three lines: 1) nat=yes 2) externip=your.external.IPaddess (or externhost=your.external.hostname) 3) localnet=192.168.0.0/24 (assuming your network uses 192.168.0.x addresses). Then "asterisk -rx sip reload" at the CLI. See ?? ports and ?? rtp for port forwarding information. |
| 19:01.18 | [TK]D-Fender | ntbourey: So do you get audio direct from * to an individual softphone both ways? |
| 19:01.50 | ntbourey | [TK]: I'm not sure |
| 19:02.06 | ntbourey | [TK]: I think I have it set up to a B2B |
| 19:02.15 | ntbourey | At least based on what the books says |
| 19:02.41 | [TK]D-Fender | ntbourey: what "it"? |
| 19:02.52 | [TK]D-Fender | ntbourey: And what are you running for clients? |
| 19:03.04 | ntbourey | A mac app called Telephone |
| 19:03.15 | ntbourey | and it refers to Asterisk |
| 19:03.27 | joseph__ | [TK]D-Fender do you have an idea on how can i solve my hitch |
| 19:05.16 | foo | ntbourey: hm, I'm currently using trixbox, asterisk under the hood. I have sip clients internally and one external. The external one is timing out. |
| 19:05.25 | foo | [TK]D-Fender: oh, thanks, missed that |
| 19:05.32 | ntbourey | foo: Okay |
| 19:05.32 | [TK]D-Fender | joseph__: You are specifying music and early media. You certainly can't have both |
| 19:06.00 | [TK]D-Fender | foo: And yes, I KNOW you're using Trixbox which is why I linked you the guide from #freepbx, and not OURS |
| 19:06.04 | joseph__ | [TK]D-Fender can you inform me why |
| 19:06.12 | [TK]D-Fender | foo: You are better off asking for support in there. |
| 19:06.24 | [TK]D-Fender | foo: GUI's are not supported in this channel |
| 19:06.35 | Kobaz | gooey |
| 19:06.46 | *** join/#asterisk chazz (n=chazz@173-24-217-85.client.mchsi.com) |
| 19:07.00 | joseph__ | [TK]D-Fender ! |
| 19:07.05 | foo | [TK]D-Fender: I know CLI. I made those changes, thanks, testing now |
| 19:10.19 | jaytee | I had Trixbox with Asterisk "under the hood" but I ripped out Asterisk and replaced it with a 426 HEMI and added a NOS system. |
| 19:10.31 | foo | jaytee: heh |
| 19:10.42 | Qwell | jaytee: so, how is your Gentoo install doing? |
| 19:10.49 | tfrew | jaytee: supercharger? |
| 19:11.09 | Chainsaw | pigpen: I'm thinking of zaptel flags, yes. |
| 19:11.27 | jaytee | now I can go from 0 to 1-800-holycrap in less than 3 seconds |
| 19:12.06 | foo | [TK]D-Fender: hm, no bueno on adding those 3 lines to my sip_nat.conf. strange |
| 19:12.07 | jaytee | Qwell, you must have mistaken me for some other person who is into self-inflicted wounds, cutting and Gentoo. |
| 19:12.25 | tfrew | Gentoo turns Qwell on |
| 19:12.42 | ntbourey | [TK]: from the CLI: I get this when I place a call to myself: http://pastebin.com/d73a18b9c |
| 19:12.58 | Qwell | jaytee: you said you added a NOS system. Figured it was a USE flag |
| 19:13.11 | *** part/#asterisk cpoulson (n=ircfs@204.246.139.68) |
| 19:13.38 | joseph__ | [TK]D-Fender do you know from where i can edit the function where it detects the early media |
| 19:13.43 | *** join/#asterisk stochastik (n=ircfs@204.246.139.68) |
| 19:13.46 | jaytee | Qwell, LOL |
| 19:13.51 | pigpen | Chainsaw, I looked at the package, it has "" for the "IUSE" |
| 19:14.05 | Chainsaw | pigpen: Yeah, as I said I was thinking of zaptel flags. |
| 19:14.14 | stochastik | Is meetme talker detection broke in 1.6? |
| 19:14.17 | pigpen | ah, ok, so no use flags. |
| 19:14.35 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:14.48 | pigpen | really, I am not a moron when it comes to this. I have no clue why this is not enabling the echocancel |
| 19:15.35 | *** join/#asterisk smash- (n=smash@173-11-0-109-oregon.hfc.comcastbusiness.net) |
| 19:16.31 | pigpen | I guess I'll get pfeifer to assist the issue to ensure all the code is being compiled. |
| 19:16.34 | Chainsaw | pigpen: Mind filing me a bug so I can look at this Tuesday? |
| 19:16.48 | pigpen | sure... |
| 19:16.54 | smash- | anyone here a sip trunk provider? |
| 19:17.08 | *** join/#asterisk wilsonj (n=jeremy@unaffiliated/dethstar) |
| 19:19.01 | stochastik | Try Flowroute |
| 19:19.03 | pigpen | Chainsaw, http://pastebin.com/m2982adba |
| 19:19.22 | pigpen | if you need anything else, let me know. |
| 19:20.08 | pigpen | thanks for the help. |
| 19:20.12 | Chainsaw | pigpen: What I meant was, could you file a bug on bugs.gentoo.org against net-misc/dahdi as that will end up with me. |
| 19:20.24 | pigpen | heh...yeah |
| 19:20.31 | Chainsaw | pigpen: Then I can take my Asterisk 1.6 test box and dissect the ebuild, see what I missed. |
| 19:20.41 | pigpen | if you missed. |
| 19:20.46 | pigpen | k, I'll file a bug |
| 19:20.55 | Chainsaw | Thanks. |
| 19:21.13 | pigpen | I should have don this in the first place, Pfeifer has told me enough times. |
| 19:21.14 | pigpen | :) |
| 19:22.21 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 19:22.27 | VaGoNeTaS | hello |
| 19:22.44 | VaGoNeTaS | does anyone knows how to convert an gsm file into an mp3 and vice versa? |
| 19:23.00 | VaGoNeTaS | or any program to do it? |
| 19:23.16 | UQlev | VaGoNeTaS, why do you meed to convert? |
| 19:23.24 | UQlev | just to listen? |
| 19:24.16 | VaGoNeTaS | well, yes, but i have an mp3 that i have to convert it into an gsm file in oder to put it on asterisk |
| 19:24.19 | VaGoNeTaS | or as an asterisk ivr |
| 19:24.51 | UQlev | asterisk can play mp3 |
| 19:26.10 | VaGoNeTaS | so is not necessary for me to convert the mp3 into an gsm |
| 19:26.56 | UQlev | VaGoNeTaS, it make sense if you want to reduce bandwidth |
| 19:27.57 | [TK]D-Fender | ntbourey: What does it look like you're calling yourself? |
| 19:28.03 | [TK]D-Fender | why* |
| 19:28.09 | ntbourey | Because I am |
| 19:28.17 | VaGoNeTaS | well, the mp3 is 689k and gsm 2,5k |
| 19:28.24 | [TK]D-Fender | ntbourey: you know i'll trust that far less than others |
| 19:28.52 | ntbourey | I already tried calling my boss but it didn't work either |
| 19:29.06 | ntbourey | So while he is busy I am using myself as a guinea pig |
| 19:29.25 | ntbourey | also I have an extension set up in my dialplan that uses an AGI script and that works okay |
| 19:32.03 | pigpen | Chainsaw, http://bugs.gentoo.org/show_bug.cgi?id=268652 |
| 19:32.33 | pigpen | I hope I did it all right, I don't do these very often. If you need more info, I'll provide anything I can. |
| 19:32.35 | Chainsaw | pigpen: Thank you, now assigned to me. |
| 19:32.48 | pigpen | tks. |
| 19:33.14 | Chainsaw | pigpen: I'll have access to test hardware tomorrow, I'll be in touch. |
| 19:33.34 | pigpen | sounds good, I'll be around. |
| 19:35.54 | UQlev | VaGoNeTaS, gentoo has in portage gsm and gsmlib |
| 19:36.57 | VaGoNeTaS | where the gsm file is defined? |
| 19:37.07 | VaGoNeTaS | in the extensions.conf file? |
| 19:42.24 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:43.42 | *** join/#asterisk yo-mama (n=bsumrall@ftnco.com) |
| 19:44.05 | Ziaeon | How do you shut up error messages such as "Asterisk command not understood" and the other random shit I get (very infrequently) at the top of the page in the recordings page? |
| 19:44.34 | eppigy | say what |
| 19:45.02 | jaytee | what |
| 19:45.08 | Ziaeon | I guess it's more of a FreePBX question, lol. |
| 19:46.01 | Ziaeon | FreePBX comes with a recordings page which is a php login for user extensions to check voicemail and call logs and what not. I realize now it's a separate product so I'll have to ask elsewhere. |
| 19:46.04 | yo-mama | sms question!!! I am a returning asterisk user hoping to get pointed in the right direction on the esiest way to get trixbox to be able to send the same text message to 100 different DIDs? |
| 19:46.09 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 19:48.35 | ruben23 | hi anyone help |
| 19:48.45 | ruben23 | <PROTECTED> |
| 19:49.11 | ruben23 | i got centos box nated and behind it is my asterisk server |
| 19:49.32 | ruben23 | my Sip voip is unreachable |
| 19:51.18 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 19:51.53 | *** part/#asterisk ntbourey (n=ntbourey@c-76-110-3-120.hsd1.fl.comcast.net) |
| 19:51.56 | ayeso | If i want to look at the comedian mail source code, do I just download the source for asterisk or is there separate code I need to find somewhere? |
| 19:52.20 | *** join/#asterisk wilsonj (n=jeremy@unaffiliated/dethstar) |
| 19:52.25 | [TK]D-Fender | ayeso: app_voicemail.c in the source tarball |
| 19:52.47 | ayeso | [TK]D-Fender: Thx, u tha man |
| 19:53.52 | [TK]D-Fender | ruben23: -A PREROUTING -p udp -m udp --dport 5060 -j DNAT --to-destination 192.168.2.3:5060 |
| 19:54.11 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 19:54.12 | [TK]D-Fender | ruben23: Why are you DNAT-ing SIP on your * box? You should not be forwarding it |
| 19:54.29 | nny_1 | is playing around with openfire XMPP and Asterisk-IM |
| 19:54.54 | nny_1 | anyone tried it or anything? It seems to work, although still working out some small issues with presence |
| 19:54.54 | [TK]D-Fender | ruben23: Or have you put another system in front of * now? |
| 19:56.33 | ruben23 | yes.. |
| 19:56.49 | ruben23 | linus box gateway===>asterisk server |
| 19:57.02 | tfrew | ruben23: is that the router i build for you a while ago? |
| 19:57.18 | ruben23 | internet===>gateway linux box====>asterisk server |
| 19:57.48 | ruben23 | tfrew...cant remember |
| 19:57.56 | ruben23 | but i ask help here.. |
| 19:57.59 | ruben23 | now |
| 19:58.03 | ayeso | [TK]D-Fender: you seem to know alot about most everything asterisk related, how hard do you think it would be to change the menu options in comedian mail, that is if you press 7 to delete a message now, to change it to 8 or something? |
| 19:58.12 | yo-mama | anyone know of a sms modual for asterisk? can point me to a good sms for asterisk howto? |
| 19:58.41 | ruben23 | my asterisk server says unreachable to my voip carrier Ip |
| 19:59.36 | [TK]D-Fender | ayeso: You're talking about changing * sourse. You didn't seem to know where to find it so it hink you'd better get familiar first |
| 19:59.45 | yo-mama | ruben23: ping your carrier from the asterisk box and then do an nmap from outside your network to your asterisk box. |
| 20:00.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:01.54 | [TK]D-Fender | ruben23: -A PREROUTING -p tcp -m tcp --dport 4569 -j DNAT --to-destination 192.168.2.3:4569 |
| 20:01.59 | [TK]D-Fender | ruben23: FYI, IAX2 = UDP |
| 20:02.11 | yo-mama | ruben23: nmap should show that port 5060 is reachable from outside your network and you should be able to ping your carrier from your asterisk box. if either of these two are wrong, you know what you need to do. If both are good, call your carrier because the issue is more than likely on there side. |
| 20:02.15 | [TK]D-Fender | ruben23: And you should take this up with 33linux ro ##networking |
| 20:05.17 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 20:07.01 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 20:14.06 | *** join/#asterisk jtodd (n=jtodd@dslb-088-072-212-085.pools.arcor-ip.net) |
| 20:14.06 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 20:15.10 | stochastik | What's the proper method to place a bounty on a bugfix? |
| 20:16.37 | [TK]D-Fender | stochastik: What bug? |
| 20:16.56 | Qwell | 15031 I'm guessing |
| 20:17.04 | stochastik | ja |
| 20:17.15 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:17.21 | stochastik | Even if it's not really a "bug"... I just want to resolve it. |
| 20:18.01 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 20:18.22 | [TK]D-Fender | stochastik: Still doesn't tell me anything |
| 20:18.37 | [TK]D-Fender | Qwell: Got a link? |
| 20:18.53 | stochastik | [TK]D-Fender: http://bugs.digium.com/view.php?id=15031 |
| 20:19.35 | Qwell | stochastik: set the o option |
| 20:19.51 | Qwell | Why? I don't know, but it's probably wrong. |
| 20:20.14 | stochastik | Set the o and T together, right? |
| 20:20.17 | Qwell | yes |
| 20:20.54 | stochastik | ahh... there's the talking events, but now no audio is being sent |
| 20:21.01 | Qwell | yeah... |
| 20:21.22 | stochastik | I suppose it has something to do with a threshold for the o option? |
| 20:29.15 | crevetor | Does anyone know why peers could show up as lagged then unreachable then come back even though there are no network problems |
| 20:34.22 | Qwell | stochastik: see issue 13801 |
| 20:34.59 | Qwell | if it works, feel free to donate the bounty amount to FreeNode or the FSF. (though you're under no obligation to do so) |
| 20:35.24 | stochastik | Qwell: I'll have a look... thanks! |
| 20:35.51 | Qwell | maybe I should have tried compiling first... hopefully it works |
| 20:38.19 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-3d3e6e54b4c17189) |
| 20:38.20 | *** part/#asterisk nealix (n=np20433@nat/sun/x-d2b4543db217d42e) |
| 20:44.06 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 20:46.07 | *** part/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 20:46.42 | stochastik | Compiles and appears to work like a charm... thanks... I'll send a donation to Freenode. |
| 20:48.13 | *** join/#asterisk madsara (i=madsara@2001:328:2002:f159:0:0:0:1) |
| 20:48.40 | madsara | This is odd - should I be receiving OPTIONS requests from my SIP provider every 3 seconds? |
| 20:48.43 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 20:48.49 | a1fa | anyway to have multiple bindports for sip? |
| 20:48.58 | madsara | It just seems a bit odd. |
| 20:49.11 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 20:51.52 | a1fa | argh |
| 20:52.08 | a1fa | i am trying to figureout why I am getting unautorrized access on my phone |
| 20:52.13 | a1fa | everything looks correct |
| 20:52.27 | a1fa | could it be that the sip phone is md5suming the password to authenticate? |
| 20:52.39 | KyleK | do toll free numbers get called by telemarketers? |
| 20:52.50 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 20:53.54 | *** join/#asterisk ta^3 (n=tacvbo@189.137.13.198) |
| 20:56.03 | nny_1 | eh FYI so far openfire + asterisk = garbage ha |
| 20:57.15 | nny_1 | it looks like the whole thing is just an elaborate scheme to try and sell the non-gpl version, i could be a bit cynical, but the whole gpl side looks abandoned |
| 20:57.27 | *** join/#asterisk youngproguru (n=quassel@74.10.229.45) |
| 20:59.03 | a1fa | this is weerd |
| 20:59.24 | a1fa | Asterisk SIP is 2.0 compliant, right? |
| 20:59.36 | KyleK | a1fa: is it unauthorized then authorized? |
| 21:00.26 | a1fa | first packet is Register |
| 21:00.39 | a1fa | second packet is rply from server: 100 TRYING |
| 21:00.48 | a1fa | third packet is: 401 UNAUTHROIZED |
| 21:00.54 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:01.18 | a1fa | :) [TK]D-Fender > * |
| 21:02.34 | [TK]D-Fender | :( * ) |
| 21:02.40 | [TK]D-Fender | : ( * ) |
| 21:02.45 | [TK]D-Fender | Smily fail |
| 21:02.46 | a1fa | how's it hanging D |
| 21:02.55 | [TK]D-Fender | a1fa: Still breathing |
| 21:03.06 | a1fa | i think my coworker has swine flue |
| 21:03.18 | a1fa | he has been chocking on his spit for last 6h |
| 21:03.26 | a1fa | spit nasty luggies too :P |
| 21:04.05 | kn0x | asterisk not liking my odbc config |
| 21:04.29 | kn0x | says it cant connect to datasource |
| 21:04.35 | kn0x | any better way to troubleshoot thiS? |
| 21:04.37 | a1fa | ok.. so X-PRO works.. but my other sip client does not |
| 21:04.38 | a1fa | weerd |
| 21:06.08 | kn0x | cdr_odbc: Unable to connect to datasource: asterisk |
| 21:06.20 | [TK]D-Fender | Swine flu... what a crock of shit. More people die do to boring strains and raging hippo attacks. |
| 21:06.51 | a1fa | this is crazy |
| 21:06.58 | a1fa | it took 14 packets to authenticate with X-PRO |
| 21:07.04 | a1fa | and i also got 401 error |
| 21:07.12 | a1fa | then Xpro did something different and authenticated to the user |
| 21:07.15 | a1fa | to the server* |
| 21:08.40 | a1fa | [TK]D-Fender : this is crazy |
| 21:08.51 | a1fa | anyway i can get more details from asterisk on the problem? |
| 21:09.03 | a1fa | i did tcpdump and compare X-PRO and mjsip sessions |
| 21:09.12 | [TK]D-Fender | a1fa: Don't what details you do have |
| 21:09.15 | a1fa | in both sessions Server responds with 401 |
| 21:09.26 | a1fa | but X-PRO changes something and authenticates to the server anyway |
| 21:09.31 | a1fa | while mjsip does not do anything |
| 21:09.38 | a1fa | i can pastebin ASCII output ? |
| 21:09.40 | a1fa | would that help |
| 21:11.19 | kn0x | is there an easy way to test odbc is config'd properly? |
| 21:11.25 | Qwell | use it |
| 21:11.27 | a1fa | http://pastebin.ca/1412572 |
| 21:12.23 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 21:12.40 | a1fa | i dont get it |
| 21:12.41 | a1fa | :( |
| 21:12.49 | a1fa | username and password work just fine |
| 21:13.57 | a1fa | ah i see it now |
| 21:14.03 | a1fa | it never passed authentication digest |
| 21:18.44 | a1fa | looks like a known issue with sipdroid |
| 21:20.04 | Qwell | a1fa: tmobile? |
| 21:20.56 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 21:24.36 | *** join/#asterisk hepta (i=cso@78.156.12.251) |
| 21:25.51 | hepta | hello. anyone used socat udp-l:$lport udp:$raddr:$rport echo rtp? |
| 21:26.19 | hepta | ok, it wont send much rtcp. but just to get rtp running for a test .. |
| 21:28.03 | a1fa | anybody got a patch for sipdroid on * |
| 21:28.08 | *** join/#asterisk jtodd (n=jtodd@dslb-088-072-251-206.pools.arcor-ip.net) |
| 21:28.09 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:29.19 | *** part/#asterisk plq (n=plq@88.250.169.4) |
| 21:32.57 | a1fa | join #mjsip |
| 21:33.13 | Ziaeon | fail :P |
| 21:40.43 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:43.45 | *** join/#asterisk LeddyHM (n=NONE@you.cant.hack.thisbox.org) |
| 21:47.48 | kn0x | [TK]D-Fender: any idea how to trouble shoot asterisk not connecting to unixodbc datasource |
| 21:56.23 | *** join/#asterisk telecos (n=sergio@42.166.219.87.dynamic.jazztel.es) |
| 22:00.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:02.48 | *** join/#asterisk bbryant (n=bbryant@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:04.44 | *** join/#asterisk bbryant2 (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:05.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:06.15 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 22:09.32 | *** join/#asterisk Chuggs (n=tadd@s142-179-186-158.ab.hsia.telus.net) |
| 22:09.43 | *** join/#asterisk pmhaddad (n=pmhaddad@97-83-155-90.dhcp.mrqt.mi.charter.com) |
| 22:10.26 | *** join/#asterisk bbryant (n=bbryant@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:13.56 | *** join/#asterisk tainted_ (n=Administ@67.43.165.100) |
| 22:14.15 | tainted_ | hello |
| 22:15.11 | tainted_ | can anyone recommend a good voip hardphone w/ xml/xhtml browser |
| 22:15.30 | *** part/#asterisk crunge (n=Crunge@dsl093-034-021.snd1.dsl.speakeasy.net) |
| 22:18.25 | jaytee | <PROTECTED> |
| 22:20.48 | BlargMaN00 | tainted_: Polycom 670 if you like a color display... |
| 22:22.42 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 22:23.38 | jaytee | oooh! color! me want! |
| 22:24.07 | BlargMaN00 | jaytee: it's shiney too... 8)~ |
| 22:24.31 | jaytee | shiny is good! |
| 22:24.36 | generalhan | hey all, im looking into monitoring software for asterisk ... i was wondering if there was a consensus as to THE ONE to use. |
| 22:24.49 | generalhan | my boss is willing to shell out the cash for queueMetrics, but i dont know if this is just overkill. all we need is to be able to track all the calls that come in seperated by the number they dialed... and how long the call was. and that is about it. |
| 22:25.05 | tainted_ | jaytee: BlargMaN00 |
| 22:25.18 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:25.28 | jaytee | ? |
| 22:25.31 | tainted_ | is there an api for the polycoms to manipulate the xml? |
| 22:26.02 | jaytee | they have a guide for formating and a little web server applet for the phones to use |
| 22:26.28 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:26.42 | jaytee | and sample apps |
| 22:27.08 | BlargMaN00 | generalhan: are these calls going to a queue, or directly to a person?? |
| 22:27.14 | jaytee | I'm pretty sure they've even got a website. I vaguely recall visiting about several hundred times. |
| 22:27.44 | generalhan | BlargMaN00: both ... kinda. there are some that will go to a queue, and some that will just ring a few reps at the same time |
| 22:29.31 | BlargMaN00 | generalhan: hmmm... I would think that you would be able to do that with just some simple system('echo something > /logs/logfile.log') type sort of call in the dialplan useing the h extension... |
| 22:30.07 | generalhan | BlargMaN00: agreed, and i brought that up, but i think that they are interested in being able to 'see' it in realtime also. |
| 22:30.16 | BlargMaN00 | generalhan: i beleive that all the info you need is already in variables inside asterisk, you would just need to get them into a logfile... |
| 22:30.35 | generalhan | you know, just to pull up the interface and know that of the 10 reps on the phone, X many came from this number. |
| 22:31.29 | BlargMaN00 | generalhan: well, that would be as simple as writing a little PHP page that would pull the info from the log file, and display to a webpage... then put an autorefresh call in it... I would think that would be a pretty easy page to code... |
| 22:32.46 | BlargMaN00 | generalhan: that's how i would approach it logically... to me, that would be the best compromise of least money spent, and least work done... but that's just me... |
| 22:35.12 | generalhan | BlargMaN00: nah, i dont think that its just you. that makes perfect sense. im just going over my current work load to see if i have time to customize anything that he might want in the future, rather than just buying something that is overkill now, that might have all the tools he could ever want. ya know? |
| 22:35.29 | BlargMaN00 | tainted_: http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip670.html <- good place to start... 8)~ |
| 22:37.22 | BlargMaN00 | generalhan: understandable... i know i have jumped off into projects i had no business taking on at the time, but got far enough into them, where i couldn't go back and do it another way... |
| 22:37.37 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 22:37.42 | BlargMaN00 | generalhan: that can be a pain sometimes... beleive you me... |
| 22:40.25 | tainted_ | BlargMaN00: yea i saw that, but it's only idle display.. i would like to display something during a call |
| 22:43.52 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-190-200.lns10.mel4.internode.on.net) |
| 22:44.23 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-42-42.dhcp.mrqt.mi.charter.com) |
| 22:48.27 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:54.14 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 23:04.45 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 23:34.34 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 23:35.40 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 23:35.46 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 23:38.24 | juanIMP | good night everey body Im using asterisk 1.2.32 + Debian and sometimes asterisk just explote, Im getting the next messages WARNING channel.c: Channel allocation failed: Can't create alert pipe!.....Unable to allocate SIP channel structure......Unable to create RTP audio session: Too many open files, Any Ideas..Thanks a Lot |
| 23:42.26 | ltd_wk | juanIMP: sounds like the box has a lot of open files? What's the output of "sysctl fs.file-max" ? |
| 23:43.13 | juanIMP | thanks ltd_wk fs.file-max = 65536 |
| 23:47.46 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 23:48.30 | ltd_wk | what about fs.file-nr ? |
| 23:51.04 | juanIMP | ltd_wk: fs.file-nr = 2400 0 65536 |
| 23:51.59 | *** part/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:54.53 | ltd_wk | juan: doesn't look very damning |
| 23:55.16 | ltd_wk | juan: are you doing something odd in asterisk that might be causing you to run out of file handles? |
| 23:56.30 | bmoraca | exten => 01189998819991197253,1,Dial(911) ...I wonder if my customers would appreciate that one... |
| 23:56.37 | juanIMP | nope ltd_wk while im googling, in cron I wrote ***** /etc/init.d/asterisk |