| 00:07.02 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 00:09.45 | mosty | juliusspencer, yes |
| 00:13.01 | ber123 | anyone here played with the g729 codec |
| 00:13.11 | mosty | yes |
| 00:13.21 | ber123 | i just downloaded some of the precompiled binary ones |
| 00:13.27 | ber123 | and i am transcoding into 729 from ulaw |
| 00:13.40 | ber123 | and getting a lot of audio clipping and other call quality issues |
| 00:14.01 | ber123 | the codec loads fine in asterisk and its almost ok, but i want to see if these are common issues |
| 00:14.34 | ber123 | i was going to build the codec from source |
| 00:14.47 | ber123 | but i found that the diff patch provided for 729 was for an old version of intel IPP |
| 00:14.56 | ber123 | and i couldnt find that old version anywhere on intels site anymore |
| 00:15.42 | mosty | ber123, i only use digium's g729 codec |
| 00:15.42 | ber123 | im going to try loading some different binary codecs |
| 00:15.48 | ber123 | ah ok |
| 00:15.58 | ber123 | is that a seperate thing you buy when you get a digium card? |
| 00:16.07 | ber123 | i have a couple 4 port T1 digium cards |
| 00:16.15 | ber123 | didnt know if that was supposed to be bundled in the purchase price |
| 00:16.36 | mosty | it's extra, $10 usd per channel |
| 00:17.38 | EvilDeshi | anyone know of another service like ipkall that will give you a free DID number that will work with free world dialup ? |
| 00:27.30 | juliusspencer | hi mosty was your card doa? I think mine is. |
| 00:28.58 | mosty | juliusspencer, not completely doa |
| 00:29.28 | juliusspencer | mosty, how did you check yours was not working? (I can't see mine with lspci) |
| 00:30.35 | mosty | it was working, but with lots of noise |
| 00:34.37 | mosty | anyone have experience debugging pri? i have a te412p, with a green light on the port, but i get "no d-channels available" when i try to dial |
| 00:34.56 | Mavvie | mosty: does "pri show spans" show them all up? |
| 00:35.10 | Mavvie | well, the one you have hooked up :-) |
| 00:35.32 | mosty | mavvie: pri show span 1, the status line is: Status: Provisioned, Down, Active |
| 00:35.33 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 00:35.42 | Mavvie | aha, there is your issue. |
| 00:36.01 | mosty | mavvie: why is it down? and how can i fix it? |
| 00:36.05 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:36.13 | Mavvie | mosty: no idea. |
| 00:38.12 | mosty | this is where i am stuck. the line is "down" but i can't figure out what that means |
| 00:38.54 | mosty | the line works plugged into another device (pc with a dialogic card) |
| 00:45.49 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 00:47.45 | IOscanner | anyone able to compile todays svn for 1.2 trunk? |
| 00:48.35 | IOscanner | I get an error on chan_zap |
| 00:49.11 | JunK-Y | which error? |
| 00:57.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:58.15 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-242.ph.ph.cox.net) |
| 01:00.32 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:02.30 | *** join/#asterisk tomcontr3 (n=tomcontr@179-76-246-201.adsl.terra.cl) |
| 01:02.42 | tomcontr3 | does any one knows anything about /dev/tty9 ? |
| 01:05.08 | EvilDeshi | its a terminal |
| 01:05.28 | EvilDeshi | /dev/ttyx is a connection terminal |
| 01:06.00 | tomcontr3 | Im having probles with tty9 |
| 01:06.06 | EvilDeshi | how so |
| 01:06.07 | tomcontr3 | I get Permission Denied all the time |
| 01:06.18 | tomcontr3 | and for that asterisk wont start |
| 01:06.18 | EvilDeshi | can you pastbin.ca the error |
| 01:06.27 | EvilDeshi | err pastebin.ca |
| 01:06.43 | tomcontr3 | it just says cannot open `tty9' for reading: Permission denied |
| 01:07.02 | tomcontr3 | crw-rw-rw- 1 asterisk tty 4, 9 May 26 17:38 tty9 |
| 01:07.05 | EvilDeshi | type who |
| 01:07.34 | tomcontr3 | [root@ip-68-178-167-75 dev]# who |
| 01:07.36 | tomcontr3 | lanvirt9 pts/0 May 26 17:52 (179-76-246-201.adsl.terra.cl) |
| 01:09.47 | tomcontr3 | what will happend if I use ln n -s to link another tty file and create s symbolic link to a tty9? |
| 01:11.56 | tomcontr3 | r u still there? |
| 01:12.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:12.19 | *** join/#asterisk mrichmanM (n=richmanm@c-24-20-124-243.hsd1.mn.comcast.net) |
| 01:13.41 | tomcontr3 | EvilDeshi? |
| 01:17.21 | IOscanner | anyone remember how to get amd64 smp to build. I don't remember what I did to get the source to build when asterisk errors with: make[2]: *** No rule to make target `src/k6opt.s', needed by `src/k6opt.o'. Stop. |
| 01:21.10 | *** join/#asterisk flashnet (n=flashnet@c211-28-145-176.eburwd8.vic.optusnet.com.au) |
| 01:23.21 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 01:23.45 | IOscanner | any ideas? |
| 01:25.20 | JunK-Y | <PROTECTED> |
| 01:25.27 | JunK-Y | how can i be at priority 0! hummm |
| 01:25.54 | JunK-Y | IOscanner: which CFLAGS? |
| 01:26.05 | JunK-Y | for h323? |
| 01:29.18 | IOscanner | make[2]: *** No rule to make target `src/k6opt.s', needed by `src/k6opt.o'. Stop. |
| 01:29.18 | IOscanner | make[2]: Leaving directory `/usr/src/linux-2.6.16/asterisk-1.2/asterisk-1.2/codecs/gsm' |
| 01:29.18 | IOscanner | make[1]: *** [gsm/lib/libgsm.a] Error 2 |
| 01:29.19 | IOscanner | make[1]: Leaving directory `/usr/src/linux-2.6.16/asterisk-1.2/asterisk-1.2/codecs' |
| 01:29.19 | JunK-Y | i never saw that on amd64 |
| 01:29.19 | IOscanner | I have seen this before I just don't remember what I did to get around it |
| 01:29.20 | JunK-Y | paste like 3-4 lines before that. |
| 01:29.27 | IOscanner | I know if I can get it to not use the k6 it works. I just don't remember what I did. |
| 01:30.11 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:31.11 | IOscanner | http://papernapkin.org/pastebin/view/155 |
| 01:31.34 | IOscanner | I even tried with i686 |
| 01:31.44 | IOscanner | that is how I build it before, I just don't remember what I had to change |
| 01:36.16 | IOscanner | maybe proc=k8 or proc=athlon |
| 01:36.17 | IOscanner | damn it has been a while. |
| 01:36.17 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-114-89.dsl.tulsok.swbell.net) |
| 01:39.14 | IOscanner | Any ideas? |
| 01:40.35 | mosty | anyone know how to figure out why the d-channel in my E1 pri is down? |
| 01:42.47 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 01:53.01 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:53.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:01.57 | *** join/#asterisk mosty (n=mostyn@203-206-139-58.perm.iinet.net.au) |
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| 02:23.05 | mosty | how can i i debug why my e1 d-channel is down? i can turn on intense debugging on that span, but i dont understand the output |
| 02:36.01 | mosty | is there a way to probe for d-channels? |
| 02:45.55 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 02:46.25 | IOscanner | I can't seem to get Asterisk to compile on an amd64 box. I get an error about k6opt. Here is my pastebin: http://papernapkin.org/pastebin/view/157 |
| 02:46.29 | IOscanner | any ideas? |
| 02:59.52 | *** join/#asterisk Chiko_Medina (n=kristian@189.132.90.213) |
| 02:59.54 | Chiko_Medina | hi |
| 03:00.34 | Chiko_Medina | any know how call with Zap ? i have Motorola Wildcard X100P |
| 03:01.38 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:03.54 | Chiko_Medina | any know how call with Zap ? i have Motorola Wildcard X100P? |
| 03:05.35 | NovceGuru | Chiko_Medina, maybe wait a few hours between repasting |
| 03:07.57 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 03:09.32 | IOscanner | exit |
| 03:09.35 | *** part/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 03:14.37 | *** join/#asterisk m308 (n=beemer@ool-182e45d4.dyn.optonline.net) |
| 03:15.20 | m308 | how would I route a wav from festival to asterisk to make a phone call and play that wav ? |
| 03:15.51 | m308 | please dont all answer at once |
| 03:15.52 | m308 | lol |
| 03:21.49 | *** part/#asterisk m308 (n=beemer@ool-182e45d4.dyn.optonline.net) |
| 03:29.29 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:31.53 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 03:32.11 | vAd0r | can someone tell me where to get the firmware to unlock the linksys pap2 |
| 03:45.54 | *** join/#asterisk rickw (n=rickw@70.128.124.134) |
| 03:47.26 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 03:47.45 | Aces1Up | can someone explain did number forwarding to me. i guess it is kinda hard for me to understand. |
| 03:48.08 | Aces1Up | who pays for the minutes when youget a did that is forwarded to a sip / iax account? |
| 03:52.22 | mosty | ask the did provider |
| 03:52.32 | mosty | or whoever does the forwarding |
| 03:52.36 | *** join/#asterisk bmg505 (n=leon@196.209.181.208) |
| 03:54.26 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 03:56.19 | Aces1Up | mosty i see, so the forwarder charges the minutes, and they drop the call at your asterisk box right? So it this the same as origination services? |
| 03:59.03 | vAd0r | anyone know how to unlock pap2 |
| 03:59.41 | mosty | Aces1Up, it's up to the entity that forwards the call to bill for it |
| 03:59.49 | mosty | as far as i know. so ask them |
| 03:59.58 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 04:22.47 | *** join/#asterisk Chiko_Medina (n=kristian@189.132.90.213) |
| 04:22.51 | Chiko_Medina | hi |
| 04:24.14 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:28.24 | Daejeo1 | how can asterisk codec H.264? |
| 04:28.54 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:29.03 | [TK]D-Fender | Daejeo1, its patented, so unless you're going to write your own.... good luck... |
| 04:30.31 | coppice | There are only 122 patents on H.264. nothing too much :-) |
| 04:30.58 | Daejeo1 | any open codec H.264? |
| 04:31.42 | [TK]D-Fender | Daejeo1, perhaps you are hard of hearing.... |
| 04:32.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:42.51 | Daejeo1 | [TK]D-Fender: http://www.voip-info.org/wiki/view/GXV-3000 |
| 04:43.09 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:44.18 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 04:44.20 | SwK | ummm is there a problem w/ unixODBC 2.2.11 and asterisk? |
| 04:47.09 | mosty | [TK]D-Fender, is it possible that i could get a te412p configured for E1 to light up green (but have no available d-channel), when it's connected to a T1 line? |
| 04:47.35 | [TK]D-Fender | mosty, Dunno.... I wouldn't try anything so crazy :) |
| 04:47.45 | [TK]D-Fender | mosty, set your jumpers right! |
| 04:48.13 | mosty | [TK]D-Fender, the problem is i don't actually know wether the line is E1 or T1 :/ |
| 04:48.28 | [TK]D-Fender | mosty, Where is it located? |
| 04:48.42 | mosty | australia, where E1 is standard |
| 04:48.59 | mosty | but the line was plugged into an old samsung pbx port marked "PRI/T1" |
| 04:49.29 | [TK]D-Fender | mosty, Wel... go find out... |
| 04:49.53 | mosty | i rang the provider but they can't help me until tomorrow |
| 04:50.35 | mosty | i get a green light but no d-channel when configured for E1, but i haven't had any luck when configured for T1 |
| 04:51.38 | mosty | all the useful pri debugging info i could find on the web was for sangoma hardware, but this particular box is one of the few that i've built that uses a digium pri card :/ |
| 04:52.01 | shmaltz | anyone here have an adit 600 to sell? |
| 04:56.42 | shido6 | :) |
| 04:56.58 | shido6 | d channnel on channel 16 ? |
| 04:57.52 | mosty | shido6, for an E1, yes it should be |
| 04:58.48 | shido6 | have you tried all the othter channels one at a time? :) |
| 04:59.18 | mosty | i tried a few others, but not all of them |
| 04:59.31 | shido6 | 5 minutes or 24 hrs... its your time :) |
| 04:59.35 | mosty | is there any way to probe for the correct d channel? |
| 04:59.46 | SwK | asterisk -r |
| 05:02.23 | shido6 | one at time or write a perl script :) |
| 05:02.33 | shido6 | that would be cool |
| 05:02.49 | shido6 | findDchannel.pl |
| 05:05.30 | mosty | shido6, is there a quick way to tell that it's working, besides making a test call for each possibility? |
| 05:06.01 | mosty | or is there any other reason why the default d channel is down? (another setting misconfigured?) |
| 05:06.13 | shido6 | test call |
| 05:06.14 | shido6 | um |
| 05:06.17 | shido6 | do the b channels come up? |
| 05:06.46 | shido6 | asterisk will tell you that the b channels are up if you have it setup right |
| 05:06.56 | shido6 | a message for each channel |
| 05:07.15 | shido6 | then u can make a test call |
| 05:07.32 | shido6 | if you dont see the b channels coming up dont waste time with a test call |
| 05:07.46 | shido6 | but hey - u have till tomorrow :) |
| 05:07.58 | mosty | i'm flying home tonight ;) |
| 05:08.22 | shido6 | do it on de plane |
| 05:08.48 | mosty | chan_zap shows "Registered channel X, PRI Signalling signalling" for each channel |
| 05:09.02 | shido6 | erf? |
| 05:09.48 | mosty | erf? |
| 05:10.10 | shido6 | its the sound a dog makes when you're being questioned. |
| 05:10.18 | shido6 | by the dog that is |
| 05:10.19 | shido6 | :) |
| 05:10.29 | shido6 | tilt your head to the side and look in thet mirror |
| 05:10.32 | shido6 | and go "erf?" |
| 05:10.36 | shido6 | :) |
| 05:11.00 | shido6 | ok, maybe without the F |
| 05:11.31 | mosty | is that message i pasted the one you were talking about? |
| 05:11.32 | shido6 | where geographically are you? |
| 05:11.48 | mosty | australia |
| 05:11.50 | shido6 | err... where is the asterisk box? |
| 05:12.10 | mosty | the pbx is in australia also |
| 05:12.39 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 05:14.51 | shido6 | how many lines are you supposed to get with this .au telco? |
| 05:15.35 | mosty | they say they are being billed for two 30 channel and one 10 channel isdn connections |
| 05:16.33 | mosty | but one of the cables they think does 30 channels is plugged into a port marked "PRI/T1" on a samsung pbx, which also has a disabled port labelled "E1" |
| 05:16.47 | shido6 | hehe |
| 05:17.01 | shido6 | well what do you think about that. |
| 05:17.26 | *** join/#asterisk bintut (n=bintut@cm90.gamma182.maxonline.com.sg) |
| 05:17.33 | mosty | i thought it might be a t1 line, so i'm trying that now, with even less luck than when i was trying to configure it as E1 |
| 05:17.50 | shido6 | ok well... T1's dont have 30 channels |
| 05:18.03 | bintut | hello all.. |
| 05:18.04 | shido6 | does that help you? :) |
| 05:20.04 | shido6 | if the one cable that does 30 channels is plugged into a port marked pri/t1 and t1's dont handle 30 channels what do you think the problem is? :) |
| 05:20.26 | shido6 | and there is a lonely port that is disabled marked E1 |
| 05:20.42 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 05:20.52 | shido6 | there's got to be a light switch turning on somewhere... :) |
| 05:20.52 | bintut | i don't understand here.. i have 2 sip extensions.. 1 is connected from the internet and 1 is connected from the lan.. when these 2 sip softphones talk with each other, the sip software from the lan hears it's own voice (echo) while the sip phone from the internet is clear enough without an echo.. this also applies when my analog phone connected to my fxs port calls the sip softphone from the internet, the analog phone hears an echo bu |
| 05:20.53 | bintut | t the sip softphone hears clearly.. any idea? |
| 05:21.37 | mosty | shido6, i know that t1's only have 24 channels, but the people here have never actually counted how many channels they have |
| 05:24.14 | SwK | *yawn* |
| 05:24.18 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 05:24.52 | shmaltz | yay, I finaly got my slackware boot cd to detect my 3ware 9560se raid controller :) |
| 05:29.11 | Nugget | s nugget@carrera:~>sudo tw_cli /c0 show all |
| 05:29.11 | Nugget | /c0 Driver Version = 3.60.03.005 |
| 05:29.11 | Nugget | /c0 Model = 9650SE-2LP |
| 05:29.11 | Nugget | yay 3ware. :) |
| 05:29.12 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 05:32.22 | shmaltz | Nugget, that your boot device? |
| 05:32.55 | Nugget | yeah |
| 05:33.05 | shmaltz | Nugget, slackware? |
| 05:34.03 | Nugget | FreeBSD |
| 05:34.21 | shmaltz | cool |
| 05:34.22 | *** join/#asterisk axisys (n=axisys@ip68-98-146-161.dc.dc.cox.net) |
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| 05:48.53 | bintut | anyone here knows if there's a debian etch i386 repository for the asterisk-1.4.4? |
| 05:51.05 | bintut | how do you cancel echo for the sip channels? |
| 05:56.20 | [TK]D-Fender | bintut, you don't. EC is on PSTN interfaces. |
| 05:58.35 | bintut | [TK]D-Fender: so, there's no way to remove the echo when calling through sip? |
| 05:59.30 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 05:59.39 | [TK]D-Fender | bintut, its supposed to be done by the other side, and no, there isn't |
| 06:00.26 | jql | you can remove your own echo. It's hard to fix the other guy's phone |
| 06:00.41 | bintut | [TK]D-Fender: any idea what causes this problem? having an echo on a sip calls? |
| 06:01.04 | jql | bintut: do you hear your own voice echoing back, or does the other end hear themselves? |
| 06:01.10 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 06:01.35 | [TK]D-Fender | bintut, Whats on each side of the call? |
| 06:02.07 | bintut | jql: i only hears an echo but the phone from the other guy doesn't hear an echo.. does it mean that i have to configure my softphone here to cancel the echo? but i also used an analog phone that is connected to my fxs port of my digium card but i still hear an echo when calling to the other guy's sip extension |
| 06:02.41 | [TK]D-Fender | bintut, so Zaptel FXS -> SIP phone? |
| 06:02.46 | bintut | jql: i'm the one that hears an echo.. either using my softphone (openwengo) or my analog phone connected to the fxs port |
| 06:03.14 | [TK]D-Fender | bintut, the zaptel side should have EC enabled. |
| 06:03.21 | jql | bintut: If you hear an echo, it's the other side "causing" it. They are sending your audio back to you. Does that make sense? |
| 06:03.30 | bintut | [TK]D-Fender: i have openwengo as a softphone on my desktop and an analog phone connected to the fxs port of my digium card dev kit |
| 06:04.16 | bintut | jql: i'm not sure.. sorry, i'm new to asterisk.. this is the first time i installed asterisk on my box |
| 06:04.44 | [TK]D-Fender | bintut, set "echocancel=yes" for your zaptel port and restart * |
| 06:05.26 | bintut | [TK]D-Fender: it's already on my /etc/asterisk/zapata.conf |
| 06:05.51 | [TK]D-Fender | bintut, time to start playing with gains and trying other EC routines then. |
| 06:11.24 | bintut | [TK]D-Fender: no other way? |
| 06:12.18 | [TK]D-Fender | bintut, If you don't get echo between your soft-phone and any other device then its the zaptel size. Process of elimination you know.... |
| 06:12.33 | [TK]D-Fender | bintut, Zaptel EC often sucks. |
| 06:13.17 | bintut | [TK]D-Fender: ok.. i'll try to play with this.. |
| 06:20.06 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 06:27.29 | *** join/#asterisk peanutb (n=paulb@c-71-231-82-231.hsd1.or.comcast.net) |
| 06:28.10 | peanutb | ok. this may seem like a stupid question, but is it possible to get an sip # for my asterisk install? |
| 06:31.02 | [TK]D-Fender | peanutb, ... what is a "sip #"? |
| 06:37.21 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 06:39.43 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 06:39.51 | iBuMp | good morning! |
| 06:45.46 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 06:46.07 | iBuMp | lets try this again, good morning! |
| 06:46.25 | iBuMp | anyone here that has a little time to talk about codec |
| 06:46.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:46.56 | iBuMp | whois ibump |
| 06:47.06 | iBuMp | ahh its me |
| 06:47.11 | iBuMp | ;P |
| 06:47.28 | peanutb | A SIP number. So i can use an ipkall phone number. |
| 06:49.24 | iBuMp | sup peanutb |
| 06:49.25 | [TK]D-Fender | peanutb, as they say on their site you can usea service like FWD to act as the go-between or you can set up your * to accept un-auth'd calls. |
| 06:50.05 | peanutb | ? |
| 06:50.28 | iBuMp | anyone have a sec to chat about codec g729 |
| 06:51.26 | [TK]D-Fender | peanutb, go READ THE INSTRUCTIONS : http://www.ipkall.com/ |
| 06:52.02 | [TK]D-Fender | peanutb, and sign up for FWD and use that to connect via ipkall |
| 06:52.13 | [TK]D-Fender | iBuMp, just ask |
| 06:55.24 | *** join/#asterisk sigmounte (n=sigmount@90.202.64-86.rev.gaoland.net) |
| 06:55.32 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 06:57.25 | *** join/#asterisk Corydon76-home (i=nine@pdpc/supporter/sustaining/Corydon76-home) |
| 06:57.26 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:57.27 | iBuMp | [TK]D-Fender im using the g729 but am finding it giving some funny noices at first |
| 06:57.37 | iBuMp | and I am unable to check voice mails |
| 06:59.41 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-122-115.w90-36.abo.wanadoo.fr) |
| 07:06.21 | carrar | iBuMp, did you purchase g729 licenses? |
| 07:09.33 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:13.58 | iBuMp | carrar yes |
| 07:14.12 | iBuMp | sorry for the late respone, br |
| 07:14.53 | iBuMp | if your still there |
| 07:16.35 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 07:16.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:21.50 | iBuMp | if your still there |
| 07:22.04 | iBuMp | anyone have a sec to chat about codec g729 |
| 07:22.05 | JT | "you're" and no need to repeat |
| 07:24.18 | iBuMp | thanks |
| 07:24.31 | iBuMp | JT: would you happen to have a sec? |
| 07:25.35 | JT | i have a lot of seconds, just not for much specifics on g.729 |
| 07:25.48 | JT | you can call digium tech support if that's not operating correctly |
| 07:26.07 | iBuMp | on a sunday/holiday weekend? they mon-fri. |
| 07:26.59 | iBuMp | jt my biggest issue.. users cannot use default vm prompts. Codec isnt recognizing the buttons apparently |
| 07:29.09 | JT | yes you will have to call during business hours |
| 07:29.14 | JT | what do you mean buttons? |
| 07:29.48 | *** join/#asterisk Corydon76-home (i=purple@pdpc/supporter/sustaining/Corydon76-home) |
| 07:29.48 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:31.25 | sevard | he means "give sev some free PSTN termination" |
| 07:33.02 | JT | iBuMp: ? |
| 07:34.37 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 07:38.18 | carrar | iBuMp, are you using inband dtmf? |
| 07:38.48 | JT | i don't think he wants his problem solved |
| 07:39.00 | JT | doesn't respond, and doesn't fully explain what his actual problem is :) |
| 07:40.11 | Zipper_32 | Could somebody tell me why, when using Music On Hold, it starts, but immediately stops? |
| 07:40.30 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 07:41.44 | carrar | I give up? Because you are no longer on hold? |
| 07:41.47 | *** join/#asterisk Corydon76-home (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 07:41.47 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:42.45 | Zipper_32 | carrar: The user is still on hold, but no sound is being transferred. |
| 07:43.05 | carrar | exten => 55557,1,musiconhold(default) |
| 07:43.08 | carrar | that work? |
| 07:43.16 | carrar | or something like that |
| 07:43.47 | Zipper_32 | It does the same thing. Immediately stops. |
| 07:43.58 | Zipper_32 | With the user still on hold, |
| 07:43.59 | carrar | sure your file isn't corrupt |
| 07:44.15 | Zipper_32 | Yes. I've used may different mp3's |
| 07:44.28 | Zipper_32 | many* |
| 07:44.40 | carrar | what does the console have to say about that |
| 07:45.46 | Zipper_32 | <PROTECTED> |
| 07:46.00 | Zipper_32 | Separate lines of course, but that's it. Immediately after the dial command |
| 07:46.10 | Zipper_32 | or in this case, the Musiconhold() command |
| 07:47.29 | carrar | do you actually hear any music at all |
| 07:48.00 | Zipper_32 | No. No sound. |
| 07:48.14 | carrar | what format is the music file? |
| 07:48.30 | Zipper_32 | mp3. And it works just fine on other asterisk boxes. |
| 07:48.37 | carrar | 1.4.4? |
| 07:48.41 | Zipper_32 | 1.2.18 |
| 07:49.08 | carrar | Do you need to compile format_mp3 |
| 07:49.22 | carrar | in addons |
| 07:49.23 | Zipper_32 | I've compiled it with asterisk_addons |
| 07:49.39 | carrar | module is installed? |
| 07:50.08 | Zipper_32 | yes |
| 07:50.19 | Zipper_32 | It shows up in /usr/lib/asterisk/modules/ |
| 07:51.00 | carrar | just 1 file? |
| 07:51.01 | carrar | (music file) |
| 07:51.04 | Zipper_32 | Correct. |
| 07:51.15 | carrar | whats the size on it |
| 07:51.45 | Zipper_32 | 3244160 |
| 07:51.51 | Zipper_32 | 3.2mb. |
| 07:52.03 | carrar | So you moved the original default music files out the dir? |
| 07:52.14 | Zipper_32 | Removed them, yes. |
| 07:52.21 | carrar | restarted etc... |
| 07:52.26 | Zipper_32 | Yes, and even rebooted. |
| 07:52.51 | carrar | if you move them back does it do the same thing? |
| 07:52.51 | Zipper_32 | Correct. |
| 07:53.03 | Zipper_32 | It never worked with the default sound files, so I tried my own. |
| 07:53.09 | carrar | whats the screen out put with just doing the musiconhold(default) |
| 07:53.15 | Zipper_32 | Same problem, unrelated to the mp3 files it seems. |
| 07:53.49 | Zipper_32 | It's the same: Executing MusicOnHold("SIP/601-08cec720", "") in new stack Started music on hold, class 'default', on channel 'SIP/601-082d4cc0' Stopped music on hold on SIP/601-082d4cc0 |
| 07:54.36 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D4D4.access.telenet.be) |
| 07:55.35 | carrar | You have mpg123 installed? |
| 07:55.44 | Zipper_32 | Yes |
| 07:56.00 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:56.11 | Zipper_32 | I don't have it in /usr/local/bin though.... Even after installing. |
| 07:56.48 | carrar | You did make mpg123 in your asterisk dir right? |
| 07:57.01 | carrar | and make install |
| 07:57.03 | Zipper_32 | yes |
| 07:58.39 | carrar | have you made any changes to musiconhold.conf? |
| 07:58.47 | carrar | since you originally installed asterisk |
| 07:58.53 | Zipper_32 | None |
| 07:59.06 | carrar | so it's still the default settings? |
| 07:59.09 | Zipper_32 | It's the default. |
| 07:59.51 | carrar | and you mp3 file is in /var/lib/asterisk/mohmp3 |
| 08:00.39 | Zipper_32 | Yes, |
| 08:03.19 | Zipper_32 | It's a very odd problem... It's happened to an * box of mine before, but it was always solved after installing mpg123 0.59r, |
| 08:03.33 | carrar | shouldn't be using mpg123 anyways |
| 08:03.48 | carrar | looks like it's not needed in the latest 1.2 |
| 08:04.18 | Zipper_32 | That's what it says, but I've had no problems using it. |
| 08:04.40 | Zipper_32 | Until tonight, when nothing works. |
| 08:05.03 | carrar | what about just playing the file |
| 08:05.08 | carrar | does that even work |
| 08:05.14 | Zipper_32 | The file runs. |
| 08:05.31 | Zipper_32 | It plays Sweet Caroline, by Neil Diamond. |
| 08:05.40 | Zipper_32 | It's what keeps me going late at night. =) |
| 08:06.10 | carrar | Playback(mylamesong) |
| 08:06.12 | carrar | that works? |
| 08:06.26 | Zipper_32 | I'll check. One moment. |
| 08:06.33 | carrar | assuming the file is in the correct location |
| 08:07.16 | carrar | (/var/lib/asterisk/sounds/) |
| 08:07.25 | coppice | Ah, the great poet Neil Diamond, who wrote such wonderful works for the Monkees |
| 08:09.15 | Zipper_32 | What the.... |
| 08:09.22 | Zipper_32 | It's working on playback carrar. |
| 08:09.44 | carrar | thats good times |
| 08:09.50 | Zipper_32 | It is, |
| 08:10.49 | Zipper_32 | Alright, time to see what the problem is... I'll work on this for a little while and come back if I need more moral support. |
| 08:10.53 | Zipper_32 | Thanks so far carrar. |
| 08:10.56 | carrar | heh |
| 08:11.03 | carrar | I'd start over |
| 08:11.06 | carrar | heh |
| 08:11.11 | carrar | use 1.4.4 |
| 08:11.34 | Zipper_32 | Is 1.4.4 stable enough though? |
| 08:12.04 | carrar | works for me |
| 08:12.25 | Zipper_32 | Production environment? |
| 08:12.34 | carrar | 180 users |
| 08:12.40 | carrar | seems to work fine |
| 08:12.50 | Zipper_32 | Hmm, alright then |
| 08:12.52 | *** join/#asterisk Corydon76-home (i=gold@pdpc/supporter/sustaining/Corydon76-home) |
| 08:12.53 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:12.59 | Zipper_32 | Maybe I'll just get on the boat. |
| 08:13.11 | Zipper_32 | No need for mpg123 I imagine? |
| 08:13.15 | carrar | nope |
| 08:13.27 | Zipper_32 | Alright then, I'll scrap this for the time being. |
| 08:13.31 | carrar | no need for it in your version either |
| 08:14.50 | carrar | one more thing |
| 08:15.04 | carrar | put a Answer befor MusicOnHold(default) |
| 08:15.18 | carrar | try that again |
| 08:15.18 | Zipper_32 | Got that, thanks. |
| 08:16.44 | carrar | doubt it will make any difference |
| 08:16.48 | Zipper_32 | Didn't. |
| 08:17.06 | Zipper_32 | I'm going to install 1.4.4. |
| 08:17.09 | Zipper_32 | Give that a try. |
| 08:18.21 | ingenio | anyone here own a polycom 501? |
| 08:18.32 | jql | yeah |
| 08:19.32 | Zipper_32 | Yes |
| 08:22.52 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 08:23.36 | kombi | does SIP support storing phone numbers company wide? |
| 08:28.13 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 08:30.50 | mog | sip is a protocol, so no not really, however sip phones do support that as well as asterisk |
| 08:31.43 | kombi | mog: would you know about the cisco ones? |
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| 08:33.18 | mog | ciscos sip support is sub par in my opinion, ther skinny is okay, polycom and snom are my general recemondations |
| 08:34.34 | kombi | mog: that's what I keep hearing. I tamed the beast and got almost everything to work, got to find out their directory.xml format now.. |
| 08:43.03 | Zipper_32 | I agree, Polycom's are great, except for a problem with transferring calls on my IP430's. |
| 08:43.24 | ingenio | jql, Zipper_32: does your polycom do automatic voice activity detection/silence suppression? i'm looking to avoid that, and i've heard the 501s are difficult to configure for continuous full-duplex |
| 08:44.33 | Zipper_32 | ingenio: I believe it's all part of their speakerphone system, so yes, they use silence suppression to prevent feedback, echo, etc. |
| 08:44.55 | Zipper_32 | My 501's perform very well though, |
| 08:45.24 | *** join/#asterisk EvilDeshi (n=deshi@24-159-227-225.dhcp.mdsn.wi.charter.com) |
| 08:45.30 | ingenio | I'll be using handsets over speakerphones, but silence suppression will not be accepted, and I'm afraid that it may be a problem (from some random forum posts I've read) |
| 08:45.56 | ingenio | could you confirm that you have no problem disabling it for a handset? that would be very very very much appreciated :P |
| 08:46.14 | jql | sip.cfg: <VAD voice.vadEnable="0" voice.vadThresh="15"/> |
| 08:46.29 | jql | I grepped it for ya |
| 08:47.47 | EvilDeshi | jesus fucking christ |
| 08:47.54 | Zipper_32 | Where? |
| 08:47.59 | jql | yes, my son? |
| 08:47.59 | EvilDeshi | it took me the last 4 hours to figure this damn kernel mod out |
| 08:48.01 | EvilDeshi | and i got it |
| 08:48.14 | Zipper_32 | And then the power went out and you lost everything? |
| 08:48.20 | EvilDeshi | oh crap |
| 08:48.21 | EvilDeshi | sorry |
| 08:48.28 | EvilDeshi | i thought i was on my server window |
| 08:48.40 | EvilDeshi | got a lil excited there |
| 08:48.42 | EvilDeshi | compiling ztdummy |
| 08:48.57 | Zipper_32 | I wish I got that excited compiling ztdummy.... |
| 08:49.21 | EvilDeshi | well I have never had a computer good enough |
| 08:49.42 | EvilDeshi | or I couldnt get headers for the kernels I use on my vps |
| 08:50.35 | rue_mohr | so I'm going for a new idea |
| 08:51.10 | rue_mohr | when you pick up the phone it start babbeling you an interruptable directory |
| 08:51.19 | rue_mohr | "dial 9 for an outside line, for phil dial 103, for dan dial 106..." |
| 08:52.54 | rue_mohr | no fax machines or modems... |
| 09:00.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:03.48 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
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| 09:28.34 | Daejeo1 | how can i enable the image transmission (jpeg) |
| 09:28.59 | Daejeo1 | allow=jpg? |
| 09:29.25 | Daejeo1 | i want to use sendimage function |
| 09:33.12 | Daejeo1 | ping anyone? |
| 09:39.03 | *** part/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
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| 11:50.57 | *** join/#asterisk TJ` (n=tjefnet@host81-154-61-166.range81-154.btcentralplus.com) |
| 11:51.05 | TJ` | this NAT stuff is driving me crazy |
| 11:51.28 | TJ` | ive opened all the porst |
| 11:51.34 | TJ` | DMZ'd the server |
| 11:51.40 | TJ` | have given it full free acces |
| 11:51.40 | TJ` | s |
| 11:51.44 | TJ` | and still no sound |
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| 13:12.18 | DarKnesS_WolF | there is no doc telling me how to rebrand asterisk ? and what is the license key for the applications? and the functions ? |
| 13:12.34 | *** join/#asterisk shtoom (n=godson@59.93.117.118) |
| 13:13.02 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 13:18.15 | Daejeo1 | ? filte |
| 13:18.16 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:18.24 | Daejeo1 | ? flite |
| 13:18.33 | JT | DarKnesS_WolF: what are you talking about? |
| 13:19.44 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 13:20.31 | DarKnesS_WolF | JT: i'm talking about rebranind asterisk .. let us say i want to call it sherif :-) |
| 13:21.06 | file | and do what with it? |
| 13:21.39 | TJ` | DarKnesS_WolF u talking bout the gfx in freepbx? |
| 13:21.48 | TJ` | or trixbox gui? |
| 13:21.56 | DarKnesS_WolF | file: base a prdouct .. i think that is allowed with the GPL right ? |
| 13:22.09 | DarKnesS_WolF | TJ`: no i mean teh C code |
| 13:22.23 | TJ` | ud have to find anything that says Asterisk in the source |
| 13:22.28 | DarKnesS_WolF | file: i mean u can't get the source of asterisk and rename it and like a fork or so ? |
| 13:22.28 | TJ` | change it to sherriff |
| 13:22.29 | JT | DarKnesS_WolF: that's a bit of a silly question really, if you want to rebrand it, that's for you to work out, there's no tutorial on it |
| 13:22.30 | TJ` | and recompile |
| 13:22.52 | file | sure you can fork it, but you can't sell a binary only solution without getting a license from Digium |
| 13:23.13 | DarKnesS_WolF | file: but i still can sell it with the code right ? |
| 13:24.04 | JT | TJ`: no nat sound? |
| 13:24.06 | file | I'm not a licensing guy so I don't know what you can and can't do |
| 13:24.08 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 13:24.15 | TJ` | JT got it working :) |
| 13:24.19 | JT | ok |
| 13:24.21 | TJ` | ty |
| 13:24.28 | DarKnesS_WolF | JT: change it only didn't work ... it told me something about the the application and the function provieded an invaled license key ! |
| 13:24.41 | DarKnesS_WolF | ok guys will get u the error when i'm home :-s see u alter |
| 13:24.55 | *** part/#asterisk rickw (n=rickw@70.128.124.134) |
| 13:25.08 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 13:25.30 | TJ` | actually |
| 13:25.39 | TJ` | im going to post a simple NAT tutorial |
| 13:25.45 | TJ` | cos it took me 2days to fix this |
| 13:25.53 | TJ` | and all it needed was something really stupid |
| 13:26.01 | JT | if you asked the right questions here, it could've been fixed a lot quicker :) |
| 13:26.07 | TJ` | i did last night |
| 13:26.10 | TJ` | no one helped |
| 13:26.17 | TJ` | they sent me to a bunch of stupid pages |
| 13:26.22 | JT | it's the weekend..... |
| 13:26.23 | TJ` | and told me to setup vpns |
| 13:26.24 | TJ` | lol |
| 13:26.27 | TJ` | yea i guess so |
| 13:26.28 | TJ` | lol |
| 13:26.41 | JT | they probably weren't stupid |
| 13:26.53 | TJ` | the pages, most of them were bout tunneling |
| 13:26.59 | TJ` | and super complicated sip configs |
| 13:27.03 | JT | well, vpns probably aren't necessarily |
| 13:27.11 | TJ` | they werent |
| 13:28.04 | JT | maybe you got advised by a crowd of nubs :) |
| 13:28.04 | TJ` | maybe .... |
| 13:28.04 | TJ` | na but im a noob asterisk user |
| 13:28.04 | JT | you can't just ask once and expect perfect results at that time |
| 13:28.04 | TJ` | so it'll be good for other noobs |
| 13:28.04 | TJ` | i didnt |
| 13:28.04 | TJ` | just i cant find a friendly nat resource |
| 13:28.22 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 13:28.50 | stoffell_h | TJ`, i don't know what the issue was but if you could add your knowledge to the wiki on voip-info.org it'll probably help people in the future... |
| 13:33.17 | TJ` | :) |
| 13:35.27 | TJ` | anyone know a good place to get a really shitty ipphone worth wallmouting in a hallway? |
| 13:35.50 | JT | ebay.com |
| 13:35.50 | TJ` | (uk) |
| 13:36.00 | TJ` | felt that coming |
| 13:36.09 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:36.10 | JT | an old cisco :) |
| 13:36.25 | *** join/#asterisk chuckf (n=chuck@pool-71-191-38-44.washdc.fios.verizon.net) |
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| 13:41.20 | *** join/#asterisk Schnabel (n=Miranda@p54BBA8F6.dip0.t-ipconnect.de) |
| 13:41.25 | TJ` | anyone used a budgetone? |
| 13:41.40 | JT | don't use one |
| 13:41.51 | TJ` | wat u got? |
| 13:42.13 | JT | a few polycoms, ciscos, avaya |
| 13:42.21 | Schnabel | Anyone using mISDN under debian etch? I always get kernel-oopses... |
| 13:42.36 | JT | misdn is an oops ;) |
| 13:42.54 | _charly_ | Schnabel: which version of misdn? |
| 13:42.54 | Schnabel | Bristuff ain't any better. :) |
| 13:43.14 | Schnabel | _charly_: mISDN 1.1.3 |
| 13:43.17 | JT | Schnabel: i've found it to be more reliable and easier to debug |
| 13:43.22 | JT | and NT mode actually works :) |
| 13:44.06 | Schnabel | JT: But I can't get it to run any better than mISDN. |
| 13:44.30 | Schnabel | It doesn't even compile properly. |
| 13:44.31 | _charly_ | Schnabel: i have problems with 1.1.3 too, if i load the drivers directly after boot the whole system hangs |
| 13:44.38 | JT | it compiles just fine |
| 13:44.59 | _charly_ | with 1.1.1 it's no problem, even if i load 1.1.1 directly before 1.1.3 |
| 13:45.18 | _charly_ | but i haven't done more than loading the kernel modules, yet |
| 13:45.31 | Schnabel | JT: Not here. :-) |
| 13:46.11 | JT | what are you doing wrong? |
| 13:46.48 | _charly_ | btw, is the debian package maintainer of misdn somewhere on irc? |
| 13:46.58 | Schnabel | JT: I thought you'd know! |
| 13:47.20 | JT | Schnabel: i have no idea what problems you're having.. |
| 13:48.24 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 13:48.42 | Schnabel | JT: As long as you don't know about any general problems with mISDN and Etch, I'll just keep trying. |
| 13:49.08 | JT | trying what? |
| 13:50.59 | _charly_ | Schnabel: what isdn card(s) do you have? |
| 13:51.48 | Schnabel | JT: Trying to get mISDN to work. |
| 13:52.14 | JT | sounds like an uphill battle, but okay :) |
| 13:52.28 | Schnabel | _charly_: Several single-port hfc-cards. |
| 13:52.30 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-58-116.pskn.east.verizon.net) |
| 13:53.23 | _charly_ | Schnabel: have you also tried v1.1.1? |
| 13:54.50 | Schnabel | JT: Who dares, wins. |
| 13:54.58 | Schnabel | _charly_: Trying that now. |
| 13:55.09 | JT | ok, why not dare to use the better software? |
| 13:55.39 | TJ` | why the hell wont FOP work... |
| 13:58.47 | TJ` | FOP wont accept my passwords... |
| 13:58.53 | tzafrir_laptop | _charly_, I don't think it is really actively maintained |
| 13:59.11 | _charly_ | tzafrir_laptop: the debian package? |
| 13:59.25 | tzafrir_laptop | yes, the debian package. I have tried packaging misdnuser (1.1.2) recently |
| 13:59.43 | tzafrir_laptop | but still could not get asterisk to build with it |
| 14:00.01 | _charly_ | and i tried packaging misdn-kernel |
| 14:00.11 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 14:00.26 | tzafrir_laptop | The maintainer (Simon Richter) wasn't very responsive at my recent attempts |
| 14:01.30 | tzafrir_laptop | _charly_, I'd be interested to see what you made. Please send it to me, or send me a link |
| 14:01.37 | tzafrir_laptop | tzafrir.cohen@xorcom.com |
| 14:01.38 | _charly_ | tzafrir_laptop: one moment |
| 14:01.55 | tzafrir_laptop | though I'll probably get off-line soon... |
| 14:02.23 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 14:02.55 | Schnabel | _charly_: 1.1.1 seems to work. Thanks! |
| 14:03.01 | _charly_ | Schnabel: :) |
| 14:03.27 | _charly_ | misdn uses the isdn4linux bugtracker? |
| 14:04.06 | Schnabel | Dunno. |
| 14:12.15 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.70) |
| 14:12.18 | zeeesh | hi |
| 14:14.06 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 14:16.23 | _charly_ | tzafrir_laptop: i've sent the mail to you |
| 14:16.43 | tzafrir_laptop | thanks |
| 14:17.18 | _charly_ | you can build the kernel modules with module-assistant |
| 14:19.52 | tzafrir_laptop | got your mail. I don't have a time to look at it now. I'll just note that my modest stab at the misdn package was a total rewrite with no attempt to patch the kernel itself: |
| 14:20.43 | tzafrir_laptop | http://rapid.tzafrir.org.il/~tzafrir/misdn/ |
| 14:21.15 | tzafrir_laptop | but I didn't really look at your package yet (sorry, between things) |
| 14:23.34 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 14:23.58 | *** join/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 14:24.17 | TUplink_ | is there a way to GOTOIF a string is grater than 1? |
| 14:24.28 | TUplink_ | n/m thats not going to work |
| 14:24.50 | TUplink_ | i need an if statement to basicly say if somthing is set then go to this |
| 14:25.08 | _charly_ | tzafrir_laptop: i mainly used the madwifi package as template since i thought that just copying and fixing the old misdn debian dir was at least as much work. the only thing that my package doesn't what the old misdn-kernel package does is building binary module packages |
| 14:25.12 | *** part/#asterisk shtoom (n=godson@59.93.117.118) |
| 14:26.10 | tzafrir_laptop | well, if you get m-a working, then your package is working.... |
| 14:26.38 | tzafrir_laptop | I nturally used zaptel as a template... |
| 14:27.30 | _charly_ | i used madwifi because of the m-a thing (and because madwifi is the only other free package i'm using with m-a) |
| 14:30.47 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 14:33.30 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 14:36.20 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 14:37.01 | *** join/#asterisk berktr (n=canberk@212.156.213.131) |
| 14:38.52 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 14:39.00 | berktr | hello |
| 14:39.03 | berktr | i have a question |
| 14:39.13 | berktr | let's say i have 10 different carriers |
| 14:39.38 | berktr | and i want to enter their rates table to a database and make asterisk find the least cost route |
| 14:39.40 | berktr | is it possible |
| 14:40.32 | shido6 | yes but its going to cost you time or money to implement it |
| 14:40.34 | *** part/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 14:40.44 | shido6 | ppl seem to like a2billing |
| 14:41.02 | shido6 | or write your own or google LCR Asterisk billing :) |
| 14:47.04 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 14:50.22 | Schnabel | _charly_: misdn 1.1.1 works. Thanks a lot! |
| 14:51.03 | _charly_ | Schnabel: great :) |
| 14:56.26 | *** join/#asterisk cazze (n=pc@unaffiliated/kammicazze) |
| 15:10.33 | berktr | what is the best billing system ? |
| 15:17.08 | *** join/#asterisk mojojoseph (n=mojojose@cpe-76-187-221-48.tx.res.rr.com) |
| 15:20.45 | mojojoseph | howdy |
| 15:22.38 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-104-50.d4.club-internet.fr) |
| 15:26.59 | *** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com) |
| 15:28.13 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 15:29.47 | robin_sz | so ... what I want to do, is, during a call, if I decide the caller is a moron (eg call centre etc) I can press some key combination and add them the blacklist db ... is this an already solved probelm? is there already some mystical key sequence I dont know about? |
| 15:30.44 | berktr | lol |
| 15:31.08 | robin_sz | ? |
| 15:31.21 | berktr | today, in the morning |
| 15:31.35 | berktr | my phone was ringing and i was denying to wake up |
| 15:31.42 | berktr | however i woke up and answered the phone |
| 15:32.01 | berktr | it was an automated call center saying "this is a public service message bla bla |
| 15:32.15 | robin_sz | uh huh |
| 15:32.30 | berktr | i thought about the same thing, to blacklist them.. |
| 15:32.39 | robin_sz | uh huh ... |
| 15:32.50 | *** part/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 15:33.11 | mojojoseph | i'm getting the feeling robin_sz isn't interested in the story, and just wants the key sequence :-) |
| 15:39.41 | robin_sz | quite |
| 15:39.42 | mojojoseph | ;-) |
| 15:39.42 | robin_sz | apparently its *32 in *@home |
| 15:39.48 | mojojoseph | i've got a silly question for anyone who wants to entertain it |
| 15:40.24 | mojojoseph | the dialplan for SIP numbers still goes in extensions.conf? i.e., sip.conf is for the config of SIP and the identified peers only? |
| 15:41.22 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:42.17 | mojojoseph | nm, figured it out :-) |
| 15:42.35 | Daejeo1 | how can I use flite with local extension? |
| 15:43.13 | DarKnesS_WolF | file this is the error after i did change all asterisk name to something else |
| 15:43.25 | DarKnesS_WolF | [May 27 06:37:42] WARNING[24156]: loader.c:627 load_resource: Module 'res_musiconhold.so' could not be loaded. |
| 15:43.33 | DarKnesS_WolF | [May 27 06:37:42] WARNING[24156]: loader.c:589 inspect_module: Module 'res_musiconhold.so' did not provide a valid license key. |
| 15:46.41 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 15:46.52 | iBuMp | good morning.. |
| 15:47.39 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-165-26.columbus.res.rr.com) |
| 15:47.41 | crimethinker | If you say so. |
| 15:48.23 | iBuMp | yeah i hear yah.. |
| 15:48.36 | iBuMp | i wouldnt be on IRC if it was a true good morning |
| 15:49.25 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 15:50.30 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 15:51.22 | crimethinker | indeed. |
| 15:56.26 | berktr | which one is the easiest to setup billing system? please help |
| 15:56.35 | Qwell | DarKnesS_WolF: and why are you renaming it, without understanding the implications? |
| 15:56.46 | Qwell | some of the implications are legal ones |
| 15:57.14 | DarKnesS_WolF | Qwell: really !? |
| 15:57.33 | DarKnesS_WolF | Qwell: can u point me to the right direction to read ? |
| 15:58.24 | DarKnesS_WolF | Qwell: or what is illeagal ? |
| 15:59.20 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 15:59.58 | DarKnesS_WolF | berktr: astbilling |
| 16:00.06 | Qwell | DarKnesS_WolF: why are you trying to rename it in the first place? |
| 16:00.06 | DarKnesS_WolF | berktr: or i think it was ast2billing |
| 16:00.17 | Qwell | You obviously don't understand what you're doing in regards to the code |
| 16:00.33 | DarKnesS_WolF | Qwell: i thnk if i want a product to sell it i have to remove asterisk from it right ? but say it's based on it ? |
| 16:00.41 | Qwell | uhh...no |
| 16:00.44 | DarKnesS_WolF | Qwell: yes i'm not a coder :-) i'm just messing around . |
| 16:00.46 | Qwell | hire a lawyer |
| 16:01.07 | DarKnesS_WolF | Qwell: hmmm so i can sell asterisk without renameing it ? |
| 16:01.16 | Qwell | read the GPL |
| 16:01.21 | DarKnesS_WolF | not selling it .. hmm as installatio nand implmentation ? |
| 16:01.35 | DarKnesS_WolF | Qwell: hmm okay :-s |
| 16:02.10 | Qwell | also read the trademark usage policy |
| 16:02.15 | Qwell | and...again...hire a lawyer |
| 16:02.46 | DarKnesS_WolF | Qwell: thx will do :-) |
| 16:04.09 | berktr | DarKnesS_WolF, astbilling or a2billing |
| 16:04.22 | DarKnesS_WolF | a2billing |
| 16:04.31 | berktr | i see |
| 16:04.45 | berktr | is it possible to rate all the calls made over sip |
| 16:05.03 | russellb | Qwell: people just want to take take take :-p |
| 16:05.10 | berktr | without having the customer understand that i am rating the calls |
| 16:05.12 | Qwell | russellb: indeed |
| 16:05.15 | DarKnesS_WolF | russellb: no not me :P |
| 16:05.41 | DarKnesS_WolF | russellb: Qwell i just thought as long as it's GPL i can do whatever i wwant in the code thats all .. but seems i'm wrong. |
| 16:05.58 | russellb | you can change the code however you want |
| 16:06.05 | Qwell | You can - in the GPL code. There are some things, however, that are not GPL licensed |
| 16:06.17 | [TK]D-Fender | DarKnesS_WolF, thinking is best left to traqined professionals... quit qhile you're ahead :) |
| 16:06.24 | DarKnesS_WolF | Qwell: and u can pack none GPL with GPL code ? |
| 16:06.26 | Qwell | and renaming it may exclude you from doing some things |
| 16:06.26 | russellb | software licensing is complicated ... |
| 16:06.40 | DarKnesS_WolF | :( |
| 16:06.44 | [TK]D-Fender | DarKnesS_WolF, Go read the GPL, you clearly have no clue. |
| 16:06.52 | DarKnesS_WolF | [TK]D-Fender: i hate u :D |
| 16:07.06 | [TK]D-Fender | my work here is done.... |
| 16:07.07 | russellb | Qwell: but not renaming excludes you from doing other things :) |
| 16:07.17 | [TK]D-Fender | haoummmmmmmm |
| 16:07.18 | Qwell | eh? |
| 16:07.23 | Qwell | oh, right |
| 16:07.34 | russellb | Qwell: you can't distribute something called "Asterisk" that has any modifications from what we distribute |
| 16:07.47 | Qwell | non-trivial stuff :D |
| 16:07.54 | russellb | ;) |
| 16:07.58 | Qwell | or...I forget how it was worded |
| 16:08.03 | russellb | yeah, who knows |
| 16:08.14 | russellb | i think it's time I go set up a phone to control my tv |
| 16:08.25 | [TK]D-Fender | russellb, Actually being GPL, he can mod the source as redistribute so log as he includes the full source, no? |
| 16:08.42 | russellb | [TK]D-Fender: yes, but not under the name Asterisk |
| 16:08.43 | Qwell | [TK]D-Fender: sure, but he can't call it Asterisk...TM thing |
| 16:08.50 | russellb | according to our trademark policy. |
| 16:08.59 | DarKnesS_WolF | russellb: this is the idea of why i want to rename it . |
| 16:09.37 | [TK]D-Fender | Considering the thing he's wiping out IS Asterisk's name... "Mission Accomplished" |
| 16:09.37 | DarKnesS_WolF | phone to control u TV !? |
| 16:09.37 | Qwell | DarKnesS_WolF: read every license in every file |
| 16:09.37 | russellb | [TK]D-Fender: ha, yeah |
| 16:09.40 | Qwell | things such as FPM may not be allowed to be redistributed with something that isn't Asterisk |
| 16:09.44 | [TK]D-Fender | russellb, * > IR control? easy. |
| 16:09.50 | DarKnesS_WolF | Qwell: i'll :-) and i'll try to get a lawyer last thing i want to do is viloation of GPL |
| 16:09.55 | russellb | yeah, i don't have a remote control (or an IR receiver for the TV) |
| 16:10.04 | russellb | it's mythtv. |
| 16:10.05 | [TK]D-Fender | russellb, While you're at it, X-10 your place up and have it make coffee for you like mine used to! |
| 16:10.22 | russellb | it will be easy dialplan ... mythtv has an easy TCP interface for sending commands to it |
| 16:10.25 | DarKnesS_WolF | [TK]D-Fender: wow X-10 still alive !? |
| 16:11.11 | DarKnesS_WolF | anyway thx alot guys i'll be back later. |
| 16:12.13 | russellb | np.. |
| 16:12.42 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 16:18.59 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 16:19.39 | iBuMp | can anyone guide me in the proper order of codecs I should be trying to use.. GXP2000 phones Digium TDM800 |
| 16:20.18 | iBuMp | proper= most widely used with success |
| 16:20.24 | [TK]D-Fender | iBuMp, they ALL WORK. |
| 16:20.39 | [TK]D-Fender | iBuMp, If * can speak it, it WORKS. |
| 16:20.52 | [TK]D-Fender | iBuMp, Its a question of bandwidth. |
| 16:20.57 | iBuMp | [TK]D-Fender: Your statement is assumption |
| 16:21.19 | iBuMp | the network is internal and most codecs im trying dont allow the phones to do tone dialing into voicemail |
| 16:21.23 | [TK]D-Fender | iBuMp, If you're on a local LAN to your *, use ulaw/alaw depending on where you're located (ulaw in north america, alaw everywhere else). |
| 16:21.27 | iBuMp | or at least the prompting doesnt stop |
| 16:21.43 | [TK]D-Fender | iBuMp, thats not a CODEC issue, thats a DTMF mode isse |
| 16:21.49 | iBuMp | Ahh.. |
| 16:21.56 | iBuMp | ok now im getting somewhere |
| 16:22.03 | iBuMp | please explain.. |
| 16:22.09 | [TK]D-Fender | iBuMp, You should be using rfc2833 in your SIP phone and in its entry in sip.conf "dtmfmode=rfc2833" |
| 16:22.17 | iBuMp | have that |
| 16:23.29 | steliosk | iBuMp : You have to set that to the GXP2000 also. i think it default to "inband" |
| 16:23.30 | [TK]D-Fender | iBuMp, pastebin your SIP entry for that phone and the CLI output of a failed attempt at verbose 10. |
| 16:24.45 | iBuMp | phone has this : Send DTMF: in-audio via RTP (RFC2833) via SIP INFO |
| 16:24.55 | iBuMp | I have via RTP |
| 16:24.59 | iBuMp | selected.. |
| 16:25.33 | iBuMp | in pastebin i will put sip and log? |
| 16:25.47 | [TK]D-Fender | iBuMp, pastebin our sip.conf entry |
| 16:25.48 | [TK]D-Fender | ~pb |
| 16:25.50 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:27.11 | iBuMp | PB isnt working.. tells me Query failure: Can't open file: 'recent.MYI'. (errno: 145) |
| 16:27.11 | iBuMp | lock tables recent write |
| 16:28.07 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 16:28.33 | [TK]D-Fender | there are 2 links there... |
| 16:28.48 | [TK]D-Fender | www.pastebin.ca sue seems to work for ME... |
| 16:30.39 | iBuMp | http://pastebin.ca/514197 |
| 16:32.01 | [TK]D-Fender | iBuMp, asptebin your FULL phone's config please... |
| 16:32.03 | iBuMp | actually something is definately wrong with g729.. its has a robotic sounding BONK 2 or three times then audio stops after aobut 20 secs |
| 16:32.25 | iBuMp | [TK]D-Fender: in users,conf? |
| 16:32.36 | [TK]D-Fender | iBuMp, from what you SHOWED ME. |
| 16:32.52 | iBuMp | i just showed what was on the phones WEBPAGE.. |
| 16:33.12 | iBuMp | is there a better way to get the conf file from the phone so i can show you properly |
| 16:33.12 | [TK]D-Fender | iBuMp, sorry... I don't support GUI's.... |
| 16:33.19 | [TK]D-Fender | iBuMp, there are other channels for that. |
| 16:33.33 | iBuMp | [TK]D-Fender the gui is on the phone.. |
| 16:33.43 | iBuMp | the settings come from users.conf right |
| 16:33.46 | [TK]D-Fender | iBuMp, the "conf file" is not FROM THE PHONE. It is an * config file. |
| 16:34.03 | iBuMp | [TK]D-Fender: ok, forgive my misunderstanding. |
| 16:35.02 | iBuMp | http://pastebin.ca/514210 |
| 16:35.26 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 16:35.57 | iBuMp | i do not have an entry for each extension in sip.conf. All information is in users.conf on this install. |
| 16:39.49 | *** join/#asterisk cianh (n=cianh@2001:770:125:0:0:0:0:7) |
| 16:40.01 | [TK]D-Fender | iBuMp, then what were the first 3 lines of your first pastebin from? |
| 16:41.04 | iBuMp | sip.conf |
| 16:41.07 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 16:41.23 | iBuMp | ill post the complete sip.conf if you want |
| 16:41.50 | [TK]D-Fender | iBuMp, pastebin "sip show peer 20" |
| 16:43.00 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 16:43.40 | iBuMp | http://pastebin.ca/514226 |
| 16:44.50 | [TK]D-Fender | iBuMp, well * seems fine, I'd go double check your phone... |
| 16:45.14 | iBuMp | shite! |
| 16:45.39 | iBuMp | when you say check the phone.. Are you meaning its settings? |
| 16:46.10 | [TK]D-Fender | iBuMp, what else do you think I could mean? I figure it has POWER running through it already... you mean taht's not a safe assumption either? |
| 16:46.26 | iBuMp | [TK]D-Fender : HAh your fresh this morning.. |
| 16:46.31 | iBuMp | nice.. |
| 16:46.59 | iBuMp | what i mean is the all the phones are doing the same behavior.. htats why i thought it was something in * |
| 16:47.24 | iBuMp | the only difference between 20 and the other extensions is im remote, the rest are local |
| 16:47.43 | iBuMp | even when im local i hear the same funny noises then audio dies out. |
| 16:47.56 | iBuMp | and I am unable to dial into vm |
| 16:48.01 | iBuMp | unless using PCMU. |
| 16:48.03 | robin_sz | so ... what I want to do, is, during a call, if I decide the caller is a moron (eg call centre etc) I can press some key combination and add them the blacklist db ... is this an already solved probelm? is there already some mystical key sequence I dont know about? |
| 16:48.45 | robin_sz | I found a sort of solution on voip-info, but it didn;t look like it would work in a multi-line enviroment |
| 16:49.08 | [TK]D-Fender | robin_sz, Nothing exists that you don't code yourself. |
| 16:49.33 | [TK]D-Fender | robin_sz, And if they have multiple callerid, you are indeed up a creek |
| 16:49.51 | [TK]D-Fender | robin_sz, unless you filter by name and that happens to match |
| 16:49.52 | robin_sz | OK, so its not something thats been doen, thats fine, so long as I am not re-inventing that particualr wheel |
| 16:50.03 | robin_sz | by name? |
| 16:50.10 | robin_sz | CID just carries numbers |
| 16:50.24 | [TK]D-Fender | robin_sz, * is a "telephony toolkit" not a "black-list engine with other options too!" |
| 16:50.47 | robin_sz | duh |
| 16:50.51 | [TK]D-Fender | CID has name & number. Perhaps not where YOU are, but the tech generally does |
| 16:51.02 | robin_sz | UK, its just number |
| 16:51.09 | [TK]D-Fender | robin_sz, poor blokes! |
| 16:51.34 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 16:53.05 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 16:54.08 | BSD_Tech | TK do you ever trake a day off from the net ? |
| 16:54.27 | robin_sz | so, I can guess why its not included in the standard apps, so I guess I'll have to do it myself |
| 16:54.50 | iBuMp | does CLI> reload work or shoulds * be shutdown and restarted for codec changes. |
| 16:55.43 | crimethinker | CID sux anyway. Just yesterday I got a call from +1 (740) 167-4293 |
| 16:56.22 | robin_sz | crimethinker, possibly but if it saves me from just one poxy telemarketeer a second time, im up for it |
| 16:56.24 | [TK]D-Fender | BSD_Tech, I'm usually on for at least a little bit each day |
| 16:56.46 | BSD_Tech | you need to take a day or 2 off and relax |
| 16:56.54 | [TK]D-Fender | BSD_Tech, I am relaxing :) |
| 16:56.57 | BSD_Tech | I took yesterday off |
| 16:57.01 | BSD_Tech | ok |
| 16:57.10 | robin_sz | BSD_Tech, nah, larting users IS relaxing ;) |
| 16:57.20 | [TK]D-Fender | BSD_Tech, I'm multi-tasking and watching "Escape From L.A." now too... |
| 16:57.34 | [TK]D-Fender | BSD_Tech, I wasted yesterday, and todays weather sucks... |
| 16:57.44 | BSD_Tech | ahhh |
| 16:57.54 | BSD_Tech | sunny but cool here ins Calif |
| 16:58.00 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 16:58.52 | BSD_Tech | but I am still working on dialplan |
| 17:00.58 | *** part/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:01.40 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:02.25 | BSD_Tech | TK you have any goo dial code to add to the project ? |
| 17:04.14 | BSD_Tech | goo/good |
| 17:05.58 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 17:07.32 | BSD_Tech | http://pastebin.ca/514220 |
| 17:08.18 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 17:09.32 | [TK]D-Fender | BSD_Tech, 244 : s/BD/DB |
| 17:09.51 | BSD_Tech | ? |
| 17:09.53 | *** join/#asterisk Fieldy (i=BkJcY8u5@gentoo/contributor/Fieldy) |
| 17:10.00 | BSD_Tech | ok |
| 17:12.51 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 17:13.24 | BSD_Tech | its a work in progress |
| 17:13.40 | BSD_Tech | if any of you have time to add stuff please do |
| 17:13.53 | BSD_Tech | I am tring to find a place to post the project |
| 17:18.18 | *** join/#asterisk friedrich| (n=friedric@e177251052.adsl.alicedsl.de) |
| 17:22.47 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:28.24 | kombi | would anyone have a working directory.xml for cisco phones to get the format from? Little can be found on google.. |
| 17:31.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@70.158.103.10) |
| 17:32.15 | *** join/#asterisk friedrich| (n=friedric@e177242042.adsl.alicedsl.de) |
| 17:34.19 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:41.30 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-85-185.hag.east.verizon.net) |
| 17:41.55 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:41.58 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:41.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:43.09 | angryuser | i hansin321d cdr_mysql working fine with asterisj 1.4.0 i have upgraded to 1.4.4 and now i have some partial fields in database, like without name but with billsec, they have changed the structure of table? |
| 17:43.21 | angryuser | i had* |
| 17:43.53 | *** join/#asterisk flashnet (n=flashnet@190.3.37.157) |
| 17:47.57 | *** join/#asterisk friedrich| (n=friedric@e177242042.adsl.alicedsl.de) |
| 17:49.04 | angryuser | i cant find any changes in changelog, why change something that works? |
| 17:53.53 | *** join/#asterisk sh4g (n=jsb@l007090.zseriespenguins.ihost.com) |
| 17:53.56 | _VoiceMeUp_COM | yeah would be fun to have a hcangelog on web |
| 17:54.00 | _VoiceMeUp_COM | to see main changes |
| 17:54.44 | shag- | grr |
| 17:55.31 | *** join/#asterisk sh4g (n=jsb@l007090.zseriespenguins.ihost.com) |
| 17:55.51 | TJ` | i dont know whats harder |
| 17:55.57 | TJ` | installing trixbox |
| 17:56.01 | _VoiceMeUp_COM | running it |
| 17:56.02 | TJ` | or teaching my dad how to use it |
| 17:56.02 | _VoiceMeUp_COM | ;) |
| 17:56.05 | sh4g | haha |
| 17:56.10 | _VoiceMeUp_COM | i found so many flaws |
| 17:56.16 | TJ` | "I cant hear you I cant hear you!" |
| 17:56.20 | sh4g | i'd rather run asterisk + freepbx |
| 17:56.23 | _VoiceMeUp_COM | like it includes a file custom before a main or after a main one |
| 17:56.26 | _VoiceMeUp_COM | its so creapy |
| 17:56.28 | john-eman0n | anyone want to install asterisk on a AMD 4800+ X2 Dual Core pc for me? |
| 17:56.29 | john-eman0n | ;) |
| 17:56.39 | TJ` | john-eman0n were dyu live |
| 17:56.42 | john-eman0n | usa |
| 17:56.43 | john-eman0n | ohio |
| 17:56.51 | sh4g | or asterisk standalone |
| 17:56.52 | TJ` | if u buy me a plane ticket than ill do it |
| 17:56.58 | john-eman0n | uhm |
| 17:57.00 | john-eman0n | ssh? |
| 17:57.01 | sh4g | voicemeup - you're the voip carrier in mtl arent you |
| 17:57.04 | sh4g | ahaha |
| 17:57.09 | _VoiceMeUp_COM | yeah |
| 17:57.13 | sh4g | telnet, plane ticket or nothing. |
| 17:57.13 | Nugget | telnet is eeeeeeevil! |
| 17:57.18 | john-eman0n | no it isnt |
| 17:57.19 | john-eman0n | ;) |
| 17:57.22 | sh4g | heh |
| 17:57.27 | _VoiceMeUp_COM | sh4g i knwo you ? |
| 17:57.29 | john-eman0n | look man |
| 17:57.31 | sh4g | nope |
| 17:57.32 | john-eman0n | i'll pay someone |
| 17:57.32 | john-eman0n | to do it |
| 17:57.33 | sh4g | im on the amug list though |
| 17:57.34 | _VoiceMeUp_COM | oh |
| 17:57.34 | john-eman0n | ... |
| 17:57.35 | _VoiceMeUp_COM | ;) |
| 17:57.41 | _VoiceMeUp_COM | ok yeah , |
| 17:57.42 | sh4g | saw you guys around the forums |
| 17:57.47 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:57.55 | _VoiceMeUp_COM | yep my partner sits there sometimes |
| 17:58.06 | _VoiceMeUp_COM | lots of flames there so i prefer not to visit |
| 17:58.13 | shag- | heh |
| 17:58.20 | shag- | i just lurk there |
| 17:58.27 | _VoiceMeUp_COM | i should go post our new plan.. |
| 17:58.29 | shag- | see whats going on - what providers aroe around |
| 17:58.37 | [TK]D-Fender | shag-, Ditto |
| 17:58.56 | shag- | voice: what did ranges do you have |
| 17:59.00 | shag- | and do you do porting |
| 18:00.24 | shag- | hey, here's a fastball |
| 18:00.32 | shag- | anyone ever get distinctive ringing working on the mitel 5055 |
| 18:00.33 | _VoiceMeUp_COM | yes we port |
| 18:00.51 | shag- | as in alert-info (single ring vs double ring) |
| 18:00.51 | _VoiceMeUp_COM | we have 948's in the 418 and 312 in the 514 |
| 18:00.57 | _VoiceMeUp_COM | and nice 800's |
| 18:01.01 | _VoiceMeUp_COM | like 888-808-xxxx |
| 18:01.04 | shag- | sweet |
| 18:01.11 | shag- | where are your servers |
| 18:01.38 | _VoiceMeUp_COM | GT pri's tdm's to peer1/teleglobe/videotron |
| 18:01.57 | _VoiceMeUp_COM | botu 2ms to videotron cable users and 5ms to teleglobe(bel symaptico) |
| 18:02.19 | _VoiceMeUp_COM | we have racks at iweb , direct |
| 18:02.34 | shag- | better than some other providers |
| 18:02.40 | _VoiceMeUp_COM | some clients have < 1ms latency to us.. but asterisk cant show <1 llol |
| 18:02.49 | shag- | haha roger that |
| 18:02.51 | russellb | lololol |
| 18:03.01 | rudholm | llollolllool |
| 18:03.10 | shag- | heh |
| 18:03.17 | _VoiceMeUp_COM | but a trace/ping shows like 0.200ms |
| 18:03.18 | shag- | nice bots |
| 18:03.25 | _VoiceMeUp_COM | rus /rud / |
| 18:03.27 | _VoiceMeUp_COM | ? |
| 18:03.30 | *** join/#asterisk friedrich| (n=friedric@e177242042.adsl.alicedsl.de) |
| 18:04.09 | shag- | or anti-idle |
| 18:04.13 | shag- | anyway. |
| 18:04.33 | shag- | voicemeup: i've been poorly served in the past by other voip providers in the area |
| 18:04.49 | shag- | call quality was very bad in peak times |
| 18:05.04 | shag- | probably an internet traffic issue - their servers were located out-of-state |
| 18:05.13 | shag- | not naming anyone though :) |
| 18:05.25 | _VoiceMeUp_COM | hehhe |
| 18:05.30 | _VoiceMeUp_COM | yeah |
| 18:05.32 | _VoiceMeUp_COM | hear you |
| 18:05.45 | _VoiceMeUp_COM | bah whee you term ? i mean dest |
| 18:05.48 | _VoiceMeUp_COM | or prefix |
| 18:05.52 | _VoiceMeUp_COM | 514x? |
| 18:06.01 | shag- | yeah 514 |
| 18:06.13 | shag- | one had GT PRI's and the other was with telus, i think |
| 18:06.24 | _VoiceMeUp_COM | k we got tdm term to all mtl and qc and toronto etc etc |
| 18:06.29 | shag- | but for some reason their call processing servers were out in the boonies |
| 18:06.33 | _VoiceMeUp_COM | yeah lol |
| 18:06.40 | _VoiceMeUp_COM | could be that |
| 18:06.43 | *** join/#asterisk crimethinker (i=ircuser@legacy.diamond.org) |
| 18:06.49 | _VoiceMeUp_COM | you have to know how to run your stuff |
| 18:06.53 | shag- | so that made for mucho latency and jumbled audio on peak times |
| 18:06.56 | _VoiceMeUp_COM | like sperate trasncoding boxes from main ones |
| 18:06.59 | _VoiceMeUp_COM | use ser |
| 18:07.00 | _VoiceMeUp_COM | etc |
| 18:07.08 | shag- | yeah |
| 18:07.15 | _VoiceMeUp_COM | we learned hte hard way |
| 18:07.18 | shag- | i'm a traditional telco guy so i dont get most of that |
| 18:07.29 | _VoiceMeUp_COM | that IAX sucks sometimes |
| 18:07.34 | shag- | i know more about a dms100 than anything routing massive sip communications |
| 18:07.40 | _VoiceMeUp_COM | like in between versions 1.2 and 1.4.4 theres big problems |
| 18:07.47 | shag- | really? thats good to know |
| 18:08.08 | _VoiceMeUp_COM | yeah or if you hook iax -> asterisk (sip) --> aster2 (sip) --> tdm |
| 18:08.14 | _VoiceMeUp_COM | then you can have weir stuff hapening |
| 18:08.24 | shag- | i'm still running 1.2.18 - not really rushing to upgrade.. although SLA is very tempting.. hopefully i can get this to work with mitel phones |
| 18:08.43 | shag- | in terms of signalling or audio artifacts |
| 18:08.49 | _VoiceMeUp_COM | 1/4 clients that use 1.4 know what theyre doing , the rest havwe problems |
| 18:09.05 | _VoiceMeUp_COM | note *clients* |
| 18:09.10 | _VoiceMeUp_COM | not genral pop |
| 18:09.12 | [TK]D-Fender | shag-, 1.4 SLA != SLA. Its a cheap "hack" only good for those with a TINY amount of analog lines and feel like trying to "fake it" |
| 18:09.35 | _VoiceMeUp_COM | wahts 1.4 sla btw |
| 18:09.59 | shag- | yeah i heard that |
| 18:10.09 | shag- | it's not like asterisk is a norstar that you can bolt on the wall |
| 18:10.19 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 18:10.36 | shag- | d-fender: yeah i figured... given the configuration & scripting required for it to work... but yeah i want to use it specifically for that purpose |
| 18:10.41 | shag- | 2 zap channels |
| 18:10.54 | [TK]D-Fender | shag-, fugly, but you're welcome to it... |
| 18:11.24 | [TK]D-Fender | shag-, Sure you can bolt an * server to a wall... they sell cases expressly for that reason! |
| 18:11.27 | rudholm | shag-: whose DMS100s do you work on? |
| 18:11.31 | shag- | yeah, that wasn't my point |
| 18:11.32 | IOscanner | I installed asterisk on a 2 CP amd optron 280 Dual core box with 12 gigs of RAM. Everything seems to work but it will not play wav files. Any idea that would cause that. |
| 18:11.32 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 18:11.44 | shag- | d-fender: yeah you can - but most people don't know what they're doing |
| 18:12.22 | IOscanner | Also, I can install a rebuild pkg, but I can't build asterisk from source it errors with k6opts. |
| 18:13.03 | shag- | haha |
| 18:13.27 | IOscanner | seems to die on building libgsm. |
| 18:13.29 | IOscanner | ake[2]: *** No rule to make target `src/k6opt.s', needed by `src/k6opt.o'. Stop. |
| 18:13.29 | IOscanner | make[2]: Leaving directory `/usr/src/linux-2.6.16/asterisk-1.2/asterisk-1.2/codecs/gsm' |
| 18:13.30 | IOscanner | make[1]: *** [gsm/lib/libgsm.a] Error 2 |
| 18:13.53 | BSD_Tech | why use K6 |
| 18:14.02 | BSD_Tech | you should be able just to build it |
| 18:14.09 | IOscanner | I have tried everything I can think of what am I missing. I am not using K6 Asterisk keeps trying to use it. |
| 18:14.16 | IOscanner | correct |
| 18:14.45 | BSD_Tech | hmm are you sure you have all the needed packages |
| 18:14.50 | IOscanner | I build it all the time on AMD64 bit machine, but something with the optron CPU when you build it thinks it is a K6 |
| 18:14.55 | BSD_Tech | and why not move to freebsd and asterisk |
| 18:14.58 | BSD_Tech | lol |
| 18:15.14 | IOscanner | I even tried proc=athlon |
| 18:15.14 | BSD_Tech | we have 1.4.x and zaptel and libpri and the gui |
| 18:15.18 | IOscanner | i586, i686 |
| 18:15.32 | BSD_Tech | try not setting the cpu type |
| 18:15.40 | IOscanner | did that too |
| 18:15.51 | BSD_Tech | what flavor of linux |
| 18:16.00 | IOscanner | Debian |
| 18:16.09 | IOscanner | Has worked fine on all other processors |
| 18:16.12 | BSD_Tech | theres your problem your running linux |
| 18:16.14 | IOscanner | I have never had a problem building |
| 18:16.15 | BSD_Tech | heheh |
| 18:16.24 | shag- | hmm |
| 18:16.40 | BSD_Tech | do you make clean before rebuilding |
| 18:17.00 | IOscanner | I have been using the same build for sometime. I was trying to build for the Dual Optrons. |
| 18:17.05 | IOscanner | Yes I always do make clean |
| 18:17.11 | *** join/#asterisk dallo (n=gd@201.89.28.27) |
| 18:17.16 | BSD_Tech | ok and you have smp in the kernel |
| 18:17.46 | dallo | any free softphone that supports g729 ? |
| 18:17.47 | IOscanner | I have moved a build from another box i686 and it runs, but will not play wav files in the IVR |
| 18:18.26 | BSD_Tech | most of them do |
| 18:18.27 | russellb | IOscanner: edit codecs/gsm/Makefile, and comment out the two lines that contain "k6opt" with a # |
| 18:18.46 | IOscanner | I didn't think of that. |
| 18:19.08 | tzafrir_laptop | any specific reason you keep your asterisk source under a custom linux source dir? |
| 18:19.35 | tzafrir_laptop | IOscanner, also, debian has libgsm as a separate package |
| 18:19.58 | IOscanner | Yes I had it fully automated for rebuild of kernel and asterisk. |
| 18:20.04 | IOscanner | and patching for T.38 |
| 18:20.24 | tzafrir_laptop | wow. you miss the debian way big time |
| 18:20.48 | IOscanner | perfect that did it |
| 18:21.00 | IOscanner | Thanks russellb |
| 18:21.20 | russellb | IOscanner: you're welcome |
| 18:21.25 | IOscanner | No I just have encrypted FS and other things that I never run a stock kernel |
| 18:21.35 | IOscanner | I pull from Kenrel.org and patch and build |
| 18:22.20 | tzafrir_laptop | take a look at the package kernel-package and the command make-kpkg in it |
| 18:22.41 | tzafrir_laptop | Will save you a whole bunch of scripting |
| 18:22.43 | IOscanner | I did and I use that after I modify the source and prep |
| 18:23.15 | IOscanner | before I build I pull any new updates from various SVN system and merge |
| 18:23.30 | IOscanner | Thanks for your info. |
| 18:24.27 | _VoiceMeUp_COM | ricakrd |
| 18:24.30 | _VoiceMeUp_COM | rich ? |
| 18:24.44 | _VoiceMeUp_COM | you shoudl put a svn for http://pastebin.ca/508699 |
| 18:27.31 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 18:30.44 | BSD_Tech | friend co asterisk users hand me your dialpplans let me merge then to make 1 great dial plan |
| 18:30.53 | BSD_Tech | help me fill my free time |
| 18:31.42 | *** join/#asterisk friedrich| (n=friedric@e177242042.adsl.alicedsl.de) |
| 18:32.31 | *** join/#asterisk axisys (n=axisys@ip68-98-146-161.dc.dc.cox.net) |
| 18:37.19 | *** join/#asterisk friedrich| (n=friedric@e177242042.adsl.alicedsl.de) |
| 18:39.04 | *** join/#asterisk tbehrsin (n=tbehrsin@ip-62-235-224-121.dsl.scarlet.be) |
| 18:40.07 | tbehrsin | hi, i'm having a problem finding the source of an issue with SIP INFO to RFC2833 |
| 18:40.16 | tbehrsin | new to development with asterisk |
| 18:40.33 | tbehrsin | so wondering if there's anyone who can give me a few pointers |
| 18:41.11 | tzafrir_laptop | could you be more specific? |
| 18:41.53 | tzafrir_laptop | generally sip-specific stuff is done in channels/chan_sip.c but that file is *huge*. |
| 18:42.01 | tbehrsin | i have the following setup (roughly) |
| 18:42.01 | tbehrsin | Unknown UA <> Asterisk 1.4.4 (a) <> Asterisk 1.4.4 (b) |
| 18:42.18 | tbehrsin | (a) has dtmfmode=auto, rfc2833compensate =yes |
| 18:42.28 | *** join/#asterisk zapp-branigan (n=zapp-bra@84.79.33.1) |
| 18:42.33 | tbehrsin | (b) has dtmfmode=info |
| 18:43.14 | tbehrsin | i dial extension on (b) from unknown UA which uses SendDTMF(5) and then falls through to a Wait(10) |
| 18:43.17 | tzafrir_laptop | you can set dtmfmode on a per-peer/user basis, you know |
| 18:43.58 | tzafrir_laptop | so (a) would have one settings for the UA and different settings for the connection to (b) |
| 18:44.07 | zapp-branigan | hi, i have a problem in the codec_speex.so warning[374] loader.c File not found is need more files than codecs.conf ? |
| 18:44.33 | tbehrsin | the SIP INFO is sent from (b) to (a), and (b) renders the RFC2833 RTP events, however sends start events indefinately until there are further voice frames on the (a)/(b) channel |
| 18:44.52 | tbehrsin | yes... i know about that... however there seems to be a bug in asterisk as above |
| 18:45.28 | tbehrsin | it seems there are two code blocks in the voice frame section of __ast_read |
| 18:46.00 | tzafrir_laptop | if this is indeed dtmfmode=info, you should see the events in the 'sip debug' |
| 18:46.55 | tbehrsin | it is definately info being sent, as I see the packets arriving on (a) using tethereal... and also with sip debug |
| 18:47.10 | tbehrsin | when (b) is in rfc2833 there is no issue |
| 18:47.22 | tzafrir_laptop | zapp-branigan, now codec_speex.so itself is indeed in /usr/lib/asterisk/modules? |
| 18:47.33 | zapp-branigan | yes |
| 18:47.39 | zapp-branigan | is present |
| 18:47.52 | zapp-branigan | and 777 |
| 18:47.57 | tzafrir_laptop | and codecs.conf is present? |
| 18:48.01 | zapp-branigan | yes |
| 18:48.08 | tzafrir_laptop | and readable by asterisk? |
| 18:48.08 | zapp-branigan | etc/asterisk |
| 18:48.14 | zapp-branigan | yes |
| 18:48.18 | zapp-branigan | make samples |
| 18:51.18 | tbehrsin | is this the right place to discuss asterisk development? or is there another channel I should go to? |
| 18:52.56 | tbehrsin | ah |
| 18:53.04 | tbehrsin | i have my answer ;) |
| 18:53.31 | *** part/#asterisk tbehrsin (n=tbehrsin@ip-62-235-224-121.dsl.scarlet.be) |
| 18:54.07 | tzafrir_laptop | zapp-branigan, something I can think of: if you have no better idea, strace |
| 18:54.10 | BSD_Tech | http://pastebin.ca/514220 help with this project |
| 18:54.36 | *** join/#asterisk nickpiter (i=kvirc@ppp85-140-101-198.pppoe.mtu-net.ru) |
| 18:56.50 | tzafrir_laptop | what is /tmp/.mp3 ? in app-job-posting |
| 18:57.09 | tzafrir_laptop | you then delete /tmp/ .mp3 |
| 18:58.01 | tzafrir_laptop | good thing it's remmed-out |
| 18:58.55 | tzafrir_laptop | not a good idea to have predicatable file names under /tmp . Use a private directory for that. Otherwise you open the door for symlinks attacks |
| 19:02.01 | *** join/#asterisk PMantis (n=sswitzer@cpe-74-65-53-178.rochester.res.rr.com) |
| 19:03.56 | PMantis | Hello! I configured my Polycom phones to use Asterisk as the proxy. If I want to dial 613@fwd.pulver.com (as example) directly from phone, Asterisk should proxy the connection. However, I keep getting "484 Address Incomplete". Asterisk is looking for 613 inside the phone's context in domain fwd.pulver.com, instead of forwarding the connection. Can I change this? |
| 19:04.17 | BSD_Tech | its not fully writen |
| 19:04.38 | BSD_Tech | it is setup to grab from another server adn the play it and then delete it |
| 19:06.41 | BSD_Tech | asterisk is not a proxy |
| 19:07.06 | BSD_Tech | you would need ser or some other proxy sip server |
| 19:10.06 | PMantis | Hmm, ok |
| 19:10.33 | [TK]D-Fender | PMantis, Indeed, * is NOT a proxy server, it is a B2BUA. You should be passing the calls through * in that capacity like the rest of us. |
| 19:10.50 | BSD_Tech | http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy |
| 19:11.34 | PMantis | [TK]D-Fender: Well, my client is asking for that functionality: dial a sip URL from the desk phone... perhaps SER is the answer. |
| 19:12.09 | [TK]D-Fender | PMantis, how many other URI's is he planning on dialing like that ? |
| 19:12.35 | [TK]D-Fender | PMantis, Would be a PITA to dial having to enter all the alpha part on the phone... |
| 19:12.36 | PMantis | Unsure. Perhaps I'll ask. |
| 19:12.54 | BSD_Tech | just make estensions |
| 19:13.08 | PMantis | Agreed, but they're a high-tech company, and want the ability. |
| 19:13.24 | BSD_Tech | exten = 501,n,Dial(SIP/sip:123@66.212.134.192) |
| 19:13.54 | BSD_Tech | then look into ser |
| 19:14.22 | BSD_Tech | but having to dial alpha numeric is a pain |
| 19:14.55 | shag- | hehe. |
| 19:15.03 | shag- | try doing that with 3 different types of phones |
| 19:15.22 | shag- | they all have different means of entering letters |
| 19:15.36 | shag- | so i agree with bsd_tech's last statement :) |
| 19:16.12 | PMantis | BSD_Tech: I'm very familiar with the dialplan, but my client asks for this... I have to investigate. |
| 19:16.25 | [TK]D-Fender | PMantis, do it via a click-to-call ewb script using AMI Originate then. |
| 19:16.27 | shag- | ..anybody familiar with a hotel-style wake-up application for *? |
| 19:16.56 | PMantis | [TK]D-Fender: That's a good way |
| 19:16.57 | [TK]D-Fender | shag-, there's one ont he WIKI. get searching... |
| 19:16.58 | shag- | PMantis: what kind of phones does your company use? Some phones have the ability to dial from their own web interface |
| 19:17.26 | PMantis | shag-: My client is using Polycom... and I have a couple here to play with. |
| 19:17.58 | PMantis | shag-: Yeah, an AGI wakeup script on wiki... pretty neat. |
| 19:18.43 | shag- | Thanks. I found it - the reason i was asking is that the one i found was using (blech) freepbx |
| 19:19.28 | BSD_Tech | yes its writen in PHP |
| 19:19.36 | BSD_Tech | shag |
| 19:20.56 | BSD_Tech | if you want it shang pm me an I will send you a email witht he agi scripts |
| 19:21.22 | BSD_Tech | you need to install php to run them thou |
| 19:21.26 | shag- | Is it a standalone application? |
| 19:21.34 | BSD_Tech | its a gi |
| 19:21.39 | shag- | k |
| 19:21.41 | BSD_Tech | but writen in php |
| 19:22.48 | *** join/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 19:22.52 | shag- | However I found one on the wiki, is it the same one? |
| 19:27.27 | BSD_Tech | url |
| 19:27.50 | shag- | http://www.geekthing.com/~robf/asterisk/ |
| 19:28.12 | shag- | http://www.tktr.com/asteriskwakeupapplication |
| 19:28.12 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:28.14 | shag- | etc etc |
| 19:31.40 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 19:31.52 | *** join/#asterisk CVirus (n=GoD@196.219.179.62) |
| 19:32.10 | BSD_Tech | well I cant get ftom the 2nd one |
| 19:32.13 | BSD_Tech | o well |
| 19:32.22 | *** join/#asterisk tbehrsin (n=tbehrsin@ip-62-235-224-121.dsl.scarlet.be) |
| 19:33.31 | *** join/#asterisk friedrich| (n=friedric@e177242042.adsl.alicedsl.de) |
| 19:33.54 | *** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 19:35.35 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 19:37.18 | rbd | I have one asterisk box transferring a SIP call to another asterisk box. I'd like to send an application-specific ID along in this transfer (most likely in the SIP header) that the receiving end of the transfer should be able to use. any recommendations on the best way to do this? |
| 19:37.49 | rbd | maybe setting VXML_URL and splitting on SIP_HEADER(TO)? etc... |
| 19:40.02 | [TK]D-Fender | rbd, "show application sipaddheader" , "show application transfer" |
| 19:41.07 | tbehrsin | nice :) |
| 19:53.09 | CVirus | when using zaptel and udev .... I need to make install-udev and create a group named asterisk ... is this is right ? |
| 19:53.13 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 19:53.53 | sevard | Where the heck did the month soundbytes go? |
| 19:54.34 | sevard | oooh, mon-#.gsm |
| 19:54.36 | sevard | damn you |
| 20:01.51 | CVirus | when using zaptel and udev .... I need to make install-udev and create a group named asterisk ... is this is right ? |
| 20:04.12 | BSD_Tech | soundbites |
| 20:04.21 | BSD_Tech | you dont need to repeat |
| 20:04.35 | BSD_Tech | if some one knows the answer the will tell you |
| 20:08.12 | CVirus | BSD_Tech: sorry |
| 20:09.59 | *** part/#asterisk tbehrsin (n=tbehrsin@ip-62-235-224-121.dsl.scarlet.be) |
| 20:10.13 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 20:11.32 | toerkeium | hello guys, does anyone know if h323 communications is possible through NAT ? |
| 20:17.42 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 20:19.21 | *** join/#asterisk CVirus (n=GoD@196.219.179.62) |
| 20:25.59 | *** join/#asterisk CVirus (n=GoD@196.219.179.62) |
| 20:27.41 | BSD_Tech | http://pastebin.ca/514565 |
| 20:27.53 | BSD_Tech | more to come but please add what you can |
| 20:30.49 | BSD_Tech | I need more ideas |
| 20:39.27 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42) |
| 20:43.27 | *** join/#asterisk CVirus (n=GoD@196.219.179.62) |
| 20:49.41 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 20:50.28 | zapp-branigan | hi, exixt g729 for the arm procesor? who work in asterisk ? |
| 20:52.27 | *** join/#asterisk CVirus (n=GoD@196.219.179.62) |
| 20:57.03 | [TK]D-Fender | zapp-branigan, G729 is too processor intensive to wrok on ARM I suspect. |
| 20:59.18 | zapp-branigan | :) i find g729 for arm 22 Mip |
| 20:59.29 | zapp-branigan | arm 400Mip |
| 21:00.12 | zapp-branigan | but only libraries , there is any way to compile the library in the asterisk ? |
| 21:01.15 | [TK]D-Fender | zapp-branigan, thats a question for #asterisk-dev or the mailing lists, not here... |
| 21:01.38 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 21:02.26 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 21:03.37 | *** join/#asterisk CVirus (n=GoD@196.219.179.62) |
| 21:04.34 | *** join/#asterisk Bananaskin (n=Banana@81-86-102-88.dsl.pipex.com) |
| 21:05.00 | Bananaskin | hey guys need a pointer. anyone using a cisco phone and the sccp protocol ?? |
| 21:05.08 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 21:06.12 | *** join/#asterisk phocus (n=phocus@67.32.20.66) |
| 21:06.47 | phocus | what is a good softphone that also does video and Chats(im,google,yahoo) |
| 21:06.51 | *** join/#asterisk tost_ado (n=tostado@dslb-082-083-076-067.pools.arcor-ip.net) |
| 21:07.04 | tost_ado | hi |
| 21:07.44 | tost_ado | can anyone help me with a configuration question ? |
| 21:08.27 | [TK]D-Fender | phocus, X-Lite does video now (Windows), or Ekiga on *nix |
| 21:08.33 | [TK]D-Fender | ~ask |
| 21:08.40 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:08.49 | phocus | [TK]D-Fender thanks, will try it |
| 21:09.52 | tost_ado | ok. i need to specify a file from a variable, like exten => 2,n,playback(${DIR}/${NR}) |
| 21:10.25 | tost_ado | file.c: File /var/.../2 does not exist in any format |
| 21:10.33 | tost_ado | is the error |
| 21:11.05 | tost_ado | the path looks good, the file is there, but it does'nt play |
| 21:11.31 | tost_ado | file.c: Unable to open /var/cache/asterisk/test/2 (format gsm): No such file or directory |
| 21:14.54 | [TK]D-Fender | tost_ado, pastebin the CLI output of the failed attempt at verbose 10, "ls -l (folder)" to show that its there and sane, and the related dialplan. |
| 21:16.27 | phocus | is there any option in asterisk to enablethe video transfer or is it by default, because no peice of software i use will work with video but calls are fine |
| 21:16.53 | tost_ado | May 27 21:15:41 WARNING[22833]: file.c:512 ast_openstream_full: File /var/cache/asterisk/test/2 does not exist in any format |
| 21:16.53 | tost_ado | May 27 21:15:41 WARNING[22833]: file.c:824 ast_streamfile: Unable to open /var/cache/asterisk/test/2 (format gsm): No such file or directory |
| 21:16.53 | tost_ado | May 27 21:15:41 WARNING[22833]: app_playback.c:133 playback_exec: ast_streamfile failed on IAX2/85.176.109.195:4569-1 for /var/cache/asterisk/test/2 |
| 21:17.09 | [TK]D-Fender | tost_ado, ~pb |
| 21:17.12 | [TK]D-Fender | PASTEBIN. |
| 21:17.20 | tost_ado | <PROTECTED> |
| 21:17.20 | tost_ado | -rwxrwxrwx 1 root root 19041 2007-05-26 16:23 /var/cache/asterisk/test/2 |
| 21:17.35 | tzafrir_laptop | phocus, at leat in chan_sip you have to enable it |
| 21:17.46 | [TK]D-Fender | tost_ado, your file has no extension. thats the problem. |
| 21:17.46 | tost_ado | what does pastebin mean ? |
| 21:18.04 | [TK]D-Fender | tost_ado, if its GSM encoded, rename it to 2.gsm |
| 21:18.11 | [TK]D-Fender | ~pb |
| 21:18.13 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:18.22 | tost_ado | i tried it before but i will try again |
| 21:18.25 | phocus | tzafrir i dont understand |
| 21:18.34 | tzafrir_laptop | tost_ado, maybe one of the dirs along the pathis not executable to asterisk (if asterisk is running as non-root, which is a good thing) |
| 21:19.18 | tzafrir_laptop | ok, need to go now |
| 21:19.21 | [TK]D-Fender | phocus, go lookup "SIP video" on the WIKI . It'll tell you the codecs & settings you need to enable for it to work. |
| 21:21.28 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 21:22.37 | phocus | [TK]D-Fender i cant seemt o find it,you got a link haning around i can borrow? |
| 21:22.46 | [TK]D-Fender | ~wikis |
| 21:22.47 | jbot | wikis is probably http://www.voip-info.org |
| 21:24.48 | BSD_Tech | asterisk looks for sounds in /var/lib/asterisk/sounds |
| 21:24.54 | BSD_Tech | not /cache |
| 21:25.31 | [TK]D-Fender | BSD_Tech, he provided an explicit path. Thats not the issue |
| 21:28.03 | Zipper_32 | [TK]D-Fender: Regarding my transfer problem, I've posted the console information here: http://pastebin.ca/514693 To recap, When I answer a call with an IP430, and then transfer the call, the caller hears MOH while the transfer is being pressed, but once the transfer is sent, there is only silence from the original caller while the destination phone rings. |
| 21:29.37 | [TK]D-Fender | Zipper_32, "r" = EVIL. remove it. |
| 21:29.43 | [TK]D-Fender | Zipper_32, from EVERYWHERE |
| 21:32.08 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-7-225.red.bezeqint.net) |
| 21:34.46 | Zipper_32 | [TK]D-Fender: Alright, that was added after the problem started as a possibility to remedy it. It still functions the same way with or without it. |
| 21:36.05 | Zipper_32 | Updated post: http://pastebin.ca/514714 |
| 21:36.08 | [TK]D-Fender | Zipper_32, update it, and pastebin the entire chain of the the calls output from beginning to end |
| 21:36.31 | sevard | [TK]D-Fender: 'r' is evil? |
| 21:37.09 | phocus | I am lost, i have added the stuff the page said, but still no video, does anyone have any ideas what i might be missing? |
| 21:37.29 | [TK]D-Fender | phocus, pastebin your sip.conf |
| 21:37.31 | [TK]D-Fender | ~pb |
| 21:37.36 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:38.09 | [TK]D-Fender | Zipper_32, what kind of phones are 601/2/3 ? |
| 21:38.33 | Zipper_32 | All IP430's/. |
| 21:38.49 | phocus | lol, bastebin is broking |
| 21:38.53 | [TK]D-Fender | Zipper_32, no need for "t" either. pastebin their sip.conf entries |
| 21:39.06 | [TK]D-Fender | phocus, BS. Zipper_32 here jsut did *3* |
| 21:39.30 | phocus | http://pastebin.ca/514718 |
| 21:39.40 | Zipper_32 | The 't' allows for users to throw the call into parking.... Won't I need it? |
| 21:40.44 | [TK]D-Fender | phocus, Why does only *1* user have video codecs mentioned? |
| 21:41.02 | [TK]D-Fender | Zipper_32, I believe you can transfer to 700 regardless of "t" |
| 21:41.04 | Zipper_32 | [TK]D-Fender: sip.conf entries: http://pastebin.ca/514720 |
| 21:41.43 | phocus | [TK]D-Fender you have to do this for each user, ??? i am using trix box and i see the other users config |
| 21:41.46 | [TK]D-Fender | Zipper_32, ditch the progress lines too.... and is should be "mailbox=[box]@[vmcontext] |
| 21:42.06 | [TK]D-Fender | phocus, how can 2 phones use video if only ONE support it? |
| 21:42.34 | phocus | [TK]D-Fender doesnt the general tab add it for all extesions? |
| 21:42.59 | [TK]D-Fender | phocus, Go read the guides and stop doing a half-assed job. |
| 21:43.05 | [TK]D-Fender | and finally .... |
| 21:43.08 | [TK]D-Fender | ~trixbox |
| 21:43.10 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 21:44.09 | [TK]D-Fender | phocus, I'm also not about to assume you set your clients up right either. |
| 21:45.03 | BSD_Tech | if your using trixbox you should be in #trixbox |
| 21:45.13 | Zipper_32 | [TK]D-Fender: http://pastebin.ca/514724 is the updated console. Same thing after removing mailbox and progressinband lines from sip.conf |
| 21:45.18 | BSD_Tech | its not well liked |
| 21:45.25 | tost_ado | [TK]D-Fender: here is my pb: http://pastebin.ca/514725 |
| 21:45.53 | [TK]D-Fender | phocus, and line 21 of yourpastebin shows that [602] (the only other phone you are showing is OVERRIDING the [general] stuff... |
| 21:46.08 | sevard | BSD_Tech: you're |
| 21:46.26 | Zipper_32 | tost_ado: I didn't think playback needed a '.gsm' at the end of the filename.... |
| 21:46.31 | [TK]D-Fender | tost_ado, You muct NOT add the extension to your playback command. |
| 21:46.35 | [TK]D-Fender | must* |
| 21:46.41 | *** part/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 21:46.55 | [TK]D-Fender | tost_ado, but your file itself must HAVE an extension. |
| 21:47.10 | tost_ado | ok i try |
| 21:48.39 | tost_ado | hey great, that's it ! thanks, i was confused... |
| 21:48.45 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 21:51.51 | *** part/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 21:52.22 | BSD_Tech | your all fired turn in your asterisk servers your staplers and your keyboards and mice |
| 21:52.29 | Zipper_32 | rxgain=0.0 |
| 21:52.29 | Zipper_32 | txgain=0.0 |
| 21:52.31 | Zipper_32 | erm |
| 21:52.35 | Zipper_32 | Could I keep my red stapler? |
| 21:52.54 | BSD_Tech | nope only the green ones |
| 21:53.05 | Zipper_32 | Gah! |
| 21:53.10 | BSD_Tech | the red and black ones must be handed in |
| 21:53.40 | Zipper_32 | [TK]D-Fender: No change after removing mailbox and progress parts of sip.conf. |
| 21:53.52 | BSD_Tech | all leftover supplies must be returned the the supplies closet |
| 21:57.45 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-114-89.dsl.tulsok.swbell.net) |
| 21:57.45 | Zipper_32 | BSD_Tech: You can try to figure out why my Polycom 430's aren't sending ring tones when they transfer calls... |
| 21:58.23 | [TK]D-Fender | Zipper_32, add your mailbox back, just add it in the proper format I gave you. Then reload. upon failure, repastebin your sip.conf and CLI of your last attempt |
| 21:59.12 | TJ` | does anyone use voip talk? |
| 22:00.10 | *** join/#asterisk abrdeco (n=tbmjrf@c9113931.rjo.virtua.com.br) |
| 22:00.13 | BSD_Tech | I dont have a 430 to test with so no can do |
| 22:00.35 | abrdeco | Anybody knows why TRIXBOX 2.0 have the web interface too slow ? I can't see any errors in the logs of apache. |
| 22:01.13 | Strom_M | you want #trixbox |
| 22:01.16 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 22:01.23 | *** join/#asterisk jmacz (n=jmacz@190.24.98.233) |
| 22:01.36 | TJ` | abrdeco maybe ur box is slow |
| 22:01.38 | TJ` | what spec is it |
| 22:01.44 | TJ` | #trixbox is dead |
| 22:01.48 | TJ` | has been for like days |
| 22:02.13 | abrdeco | I readed anything about bug in the AAH. |
| 22:02.32 | abrdeco | The machine is P4 3.2ghz 1gb RAM with processor utilization less than 1% |
| 22:02.49 | TJ` | u accessing it on a local network? |
| 22:02.53 | abrdeco | Sure |
| 22:02.55 | Strom_M | i think that "trixbox is junk" sums it up nicely |
| 22:03.08 | abrdeco | I takes a loooong time to open the pages. |
| 22:05.31 | BSD_Tech | if your using asterisk@hom aka/trixbox take it to #trixbox |
| 22:07.36 | BSD_Tech | or #freepbx |
| 22:08.39 | Bananaskin | hey BSD_Tech |
| 22:09.16 | BSD_Tech | Bananaskin you slipper user you |
| 22:09.25 | BSD_Tech | how have you been |
| 22:09.29 | Bananaskin | lol, hows it hanging m8 |
| 22:09.47 | Bananaskin | yeah not bad, finally got my 7970 working with SCCP and asterisk |
| 22:09.59 | BSD_Tech | lucky bastard |
| 22:10.08 | tost_ado | thanx for the quick help, see ya ! |
| 22:10.22 | Bananaskin | lucky that I got a 7970 or that SCCCP works ? |
| 22:11.24 | BSD_Tech | both |
| 22:11.41 | Bananaskin | :) yeah its quite nice tbh |
| 22:11.51 | BSD_Tech | <=== wouls like toi have a cisco wifi phone |
| 22:12.02 | BSD_Tech | but they to expensive |
| 22:12.11 | Bananaskin | yeah, too right |
| 22:12.41 | Bananaskin | got myself a couple of Mitel 5235's as well, sweet audio quality and toucj screen as well |
| 22:12.56 | BSD_Tech | ok you can send one here for testing |
| 22:13.06 | Bananaskin | yeah sure..... |
| 22:13.10 | BSD_Tech | I will take good card of it |
| 22:13.20 | BSD_Tech | care |
| 22:13.30 | BSD_Tech | lol |
| 22:13.33 | BSD_Tech | no |
| 22:13.38 | BSD_Tech | your no sucker |
| 22:13.40 | Bananaskin | u still got the Pimped 501 ? |
| 22:13.57 | BSD_Tech | yeah but have a small issue with hum now |
| 22:14.02 | Bananaskin | lol |
| 22:14.10 | BSD_Tech | when I turn on the screen light |
| 22:14.12 | Bananaskin | its the invertor |
| 22:14.16 | BSD_Tech | yeah |
| 22:14.18 | Bananaskin | yeah.. |
| 22:14.45 | Bananaskin | did u touch any of the electrostatic that was exposed, when powered up... ? |
| 22:14.55 | Bananaskin | gives you a feckin dig!!! |
| 22:15.23 | BSD_Tech | no |
| 22:15.31 | Bananaskin | try it some day you are bored |
| 22:15.32 | BSD_Tech | lol |
| 22:15.33 | Bananaskin | lol |
| 22:15.58 | Bananaskin | Wehn you feel that a hair style like Leo Sayer is suitable |
| 22:16.18 | BSD_Tech | I just want a touchscreen phone to play with and test |
| 22:16.25 | BSD_Tech | you dont need both of them |
| 22:16.26 | Bananaskin | http://en.wikipedia.org/wiki/Leo_Sayer |
| 22:16.28 | BSD_Tech | lol |
| 22:16.38 | BSD_Tech | looking |
| 22:16.44 | Bananaskin | even has a built in web browser |
| 22:16.54 | Bananaskin | and stylus for us with big fingers :) |
| 22:17.10 | BSD_Tech | yeah I have one |
| 22:17.35 | BSD_Tech | how much was the unit |
| 22:17.41 | BSD_Tech | and when can I work for you |
| 22:17.45 | BSD_Tech | as a tester |
| 22:17.56 | BSD_Tech | get paid to play with thing is fun |
| 22:18.37 | Bananaskin | I got em both for around 180 US$ (90 GBP for the pair) 400-500$ phones |
| 22:19.45 | Zipper_32 | [TK]D-Fender: I've modified the mailbox to the same configuration that you mentioned. The result is here: http://pastebin.ca/514790 |
| 22:20.01 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 22:20.23 | JT | TJ`: yes, #trixbox is about as useful as trixbox |
| 22:20.30 | [TK]D-Fender | Zipper_32, SIP please... |
| 22:20.31 | BSD_Tech | lucky bastard |
| 22:20.48 | [TK]D-Fender | JT : but at least there is symetry ;) |
| 22:20.58 | JT | indeed :) |
| 22:21.47 | [TK]D-Fender | Zipper_32, and ho long do you let it try and ring? |
| 22:21.56 | Zipper_32 | [TK]D-Fender: 10 seconds + |
| 22:22.05 | [TK]D-Fender | Zipper_32, ok, sip.conf please... |
| 22:22.58 | Zipper_32 | http://pastebin.ca/514795 |
| 22:23.24 | BSD_Tech | Bana ok so 90 bucks used from you then ? |
| 22:26.12 | Bananaskin | BSD_Tech ?? doubt it |
| 22:26.35 | BSD_Tech | lol |
| 22:26.38 | BSD_Tech | O well |
| 22:26.52 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 22:29.20 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 22:29.30 | [TK]D-Fender | Zipper_32, and jsut for sanity [general] please... |
| 22:29.38 | Zipper_32 | Sorry, coming right up |
| 22:31.23 | Zipper_32 | http://pastebin.ca/514815 |
| 22:31.44 | [TK]D-Fender | Zipper_32, Ok, I'm all out of inspiration here.... |
| 22:34.36 | Zipper_32 | =( |
| 22:34.41 | Zipper_32 | I need to call Polycom... |
| 22:34.50 | Zipper_32 | It only happens with these 430's. |
| 22:35.32 | Zipper_32 | THe 501's seem to send the ringing tones perfectly. |
| 22:35.33 | BSD_Tech | what firmware and boot rom ? |
| 22:36.36 | [TK]D-Fender | Zipper_32, indeed changing firmwares for testing would not be a bad idea. |
| 22:37.24 | Zipper_32 | I have the latest firmware, 2.1.0.2708 right now. I had the 1.6.7 originally, and they didn't transfer properly, so I upgraded. |
| 22:37.28 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 22:39.56 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 22:40.49 | [TK]D-Fender | Zipper_32, 2.1.1 <- |
| 22:41.02 | Zipper_32 | Is the newest? |
| 22:41.40 | robin_sz | OK, so .. I want to make asterisk dial from my CRM application ... the variant already built into the app writes a file /var/spool/asterisk/outgoing .... is that the sanest way to do it? |
| 22:42.04 | robin_sz | its going to be a bit of a pain, because * is not on the same box as the crm app |
| 22:42.28 | [TK]D-Fender | robin_sz, Use AMI Originate instead then. |
| 22:42.35 | robin_sz | AMI? |
| 22:42.38 | [TK]D-Fender | ~ami |
| 22:42.53 | jbot | rumour has it, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 22:42.53 | sevard | LARGFGG Does anyone use python+agi? |
| 22:42.53 | robin_sz | whoami? |
| 22:44.50 | Zipper_32 | [TK]D-Fender: So if my reseller only has 2.1.0 on their server, I'm kind of hooped until they get the latest? |
| 22:45.35 | [TK]D-Fender | Zipper_32, Seems that way.... though mind you I have a pile of 430's running 2.0.3.B (one at one point on 2.1.0) that seem all to be fine. |
| 22:46.31 | Zipper_32 | So when you pickup a call with one of them, and then transfer that call to another phone of any type, the transferred party hears ringing tones? |
| 22:47.09 | Zipper_32 | Also, 1.2.x or 1.4.x? |
| 22:47.12 | robin_sz | [TK]D-Fender, that seems the right way to do it, just got to figure out how to attach to the AMI port now from this pile of PHP :( |
| 22:47.56 | [TK]D-Fender | Zipper_32, 1.2 ... hmmm COULD be a factor, but I wouldn't think so or I'm sure I'd have heard of it in here earlier. |
| 22:48.27 | robin_sz | to be fair, the actual CRM app (XRMS) is very nice, its just I'm not a PHP guy ... |
| 22:48.28 | sevard | Why the &#&@ would a script run PERFECTLY fine in bash and then just answer and hang in agi asterisk |
| 22:48.41 | Zipper_32 | O |
| 22:48.50 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 22:49.00 | Zipper_32 | I'm on 1.2 as well, which is what is bothering me. Especially since this is a new * box install. |
| 22:49.14 | JT | robin_sz: yeah, php does suck :/ |
| 22:50.30 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas06-3467423007.dial.bell.ca) |
| 22:53.04 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:53.14 | *** join/#asterisk BullWeivel (n=bullweiv@12.165.69.254) |
| 22:59.13 | BullWeivel | am i on crack?? is it as simple as installing Asterisk, configuring sip.conf and extensions.conf and you can then call from extension to extension??? (Yes I am reading the book from Oreilly :) ) |
| 22:59.25 | JT | yes |
| 22:59.59 | BullWeivel | well Asterisk is installed...now to the SIP and extensions.conf file. actually sounds like with sip.conf you can call yourself..so that would be a start :) |
| 23:00.11 | BullWeivel | when you edit the .conf files do you have to restart * ? |
| 23:02.41 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 23:02.54 | BSD_Tech | yes |
| 23:02.59 | BSD_Tech | or reload now |
| 23:03.02 | BSD_Tech | at the cli |
| 23:03.17 | BSD_Tech | but it is best to restart |
| 23:03.24 | robin_sz | so ... voicemail ... |
| 23:03.34 | BSD_Tech | over sated |
| 23:03.36 | robin_sz | which frickin number do I press to hear the next message |
| 23:03.46 | BSD_Tech | listen to the menu |
| 23:03.52 | BSD_Tech | it tells you |
| 23:03.55 | robin_sz | that annoying canadian tells you EVERYTHING except that |
| 23:04.10 | BSD_Tech | Allision |
| 23:04.21 | BSD_Tech | or did you have all the sounds redone ? |
| 23:04.41 | robin_sz | nope |
| 23:04.50 | robin_sz | its that annoying canadian allright |
| 23:05.08 | robin_sz | press 1 for this, 3 for that ... |
| 23:05.17 | robin_sz | 7 to delete .. |
| 23:05.38 | robin_sz | no mention of next message, which is presumably the most commonly needed thing |
| 23:05.52 | BSD_Tech | well allison recorded all the files |
| 23:06.09 | BSD_Tech | I know of no canadian that recoreded sound files |
| 23:06.18 | robin_sz | she is canadian |
| 23:07.56 | BSD_Tech | 6 for next message |
| 23:08.29 | robin_sz | its entirely possible I hung up long before we got to 6 |
| 23:09.15 | BSD_Tech | 5 to repeat the msg |
| 23:09.21 | Zipper_32 | 6 is next message. |
| 23:09.23 | Zipper_32 | 7 is delete. |
| 23:09.34 | robin_sz | 1 5 and 7 I got .. |
| 23:10.09 | Zipper_32 | * to exit. |
| 23:10.51 | bkw_ | haha stop dising allison |
| 23:10.54 | BSD_Tech | 7 also undeletes msgs |
| 23:10.57 | bkw_ | i'll have to call her up and tell on you |
| 23:12.08 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:12.45 | *** join/#asterisk Mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 23:13.18 | _VoiceMeUp_COM | ahah |
| 23:13.31 | BSD_Tech | call here and tell here we still have missing sound files |
| 23:13.41 | _VoiceMeUp_COM | http://pastebin.ca/514861 |
| 23:13.44 | _VoiceMeUp_COM | is your vm menu |
| 23:13.46 | _VoiceMeUp_COM | BTW |
| 23:13.47 | BSD_Tech | they never recorded all the holidays |
| 23:14.05 | _VoiceMeUp_COM | robin ? |
| 23:14.19 | BullWeivel | canadian chicks are hot!! they don't shave their underarms |
| 23:14.24 | _VoiceMeUp_COM | ? |
| 23:14.25 | BullWeivel | LOL (sory...couldnt resist) |
| 23:14.30 | _VoiceMeUp_COM | u mean french im sure |
| 23:14.31 | _VoiceMeUp_COM | lol |
| 23:14.44 | _VoiceMeUp_COM | at least in quebec city they are the cutest iv seen.. |
| 23:14.48 | BullWeivel | canadian...frech...its all the same ;) |
| 23:14.49 | _VoiceMeUp_COM | 9 1/2 out of 10 |
| 23:14.51 | _VoiceMeUp_COM | no[pe |
| 23:15.00 | _VoiceMeUp_COM | id never even think of f.. g a french |
| 23:15.02 | Zipper_32 | BullWeivel: Say that to somebody on the west coast of Canada, and they'll cut you up. |
| 23:15.03 | BullWeivel | actualy i have seen a ton of canadian chicks who are hot so. im just being a smart ass. |
| 23:15.11 | _VoiceMeUp_COM | quebec /canda is like california |
| 23:15.15 | _VoiceMeUp_COM | in terms of chicks |
| 23:15.16 | BSD_Tech | no |
| 23:15.25 | _VoiceMeUp_COM | not candien |
| 23:15.27 | _VoiceMeUp_COM | well |
| 23:15.32 | BSD_Tech | canadians and french canadians are noightin alike |
| 23:15.44 | _VoiceMeUp_COM | yeah order is BC/QC, ON,and rest |
| 23:15.53 | BSD_Tech | french canadians should go back to france |
| 23:15.54 | _VoiceMeUp_COM | bc and qc tie.. |
| 23:15.56 | BullWeivel | lol and rest. |
| 23:16.00 | _VoiceMeUp_COM | monreal should be wiped |
| 23:16.07 | _VoiceMeUp_COM | theres no more cnadians in montreal |
| 23:16.21 | _VoiceMeUp_COM | its all arab,chineese, and africans-ameri |
| 23:16.22 | BSD_Tech | Quebec needs to have a wall built around it |
| 23:16.27 | _VoiceMeUp_COM | yes |
| 23:16.28 | BSD_Tech | thne filled with water |
| 23:16.29 | _VoiceMeUp_COM | we working on it |
| 23:16.29 | Zipper_32 | and fill this wall with water? |
| 23:16.41 | _VoiceMeUp_COM | well maybe pour the water outside the wall |
| 23:16.42 | _VoiceMeUp_COM | ;) |
| 23:16.54 | _VoiceMeUp_COM | dotn kill the chicks !!! |
| 23:17.12 | BullWeivel | ok..wtf... what are the commands to see if your sip phone is registered? i know i just saw the damn commands in this book. |
| 23:17.14 | _VoiceMeUp_COM | go on any dating forum.. filter USA.. girls 22-30 |
| 23:17.14 | BSD_Tech | fricking frenchies think they deserve everything |
| 23:17.21 | _VoiceMeUp_COM | then do same and add quebec for cnada |
| 23:17.22 | BSD_Tech | sip show peers |
| 23:17.23 | _VoiceMeUp_COM | youll understand |
| 23:17.35 | _VoiceMeUp_COM | here its rare to see an DUFB |
| 23:17.36 | JT | err |
| 23:17.41 | JT | enough blatant racism, guys |
| 23:17.46 | BullWeivel | okay..thats what i tried... well i tried sip and it said command not found. |
| 23:17.46 | JT | this is an asterisk channel |
| 23:17.46 | Zipper_32 | [TK]D-Fender: Just FYI, 2.0.3B has the same issue on my box. |
| 23:18.09 | Zipper_32 | [TK]D-Fender: Oh, and I've tried on 2 different * boxes as well... Same results. |
| 23:18.26 | BSD_Tech | exten = 30,s,1,Goto(built wall around Quebec&fille with water) |
| 23:18.36 | JT | ... |
| 23:18.37 | BSD_Tech | exten = 30,,1,Goto(built wall around Quebec&fille with water) |
| 23:18.40 | BSD_Tech | lol |
| 23:18.44 | Zipper_32 | BullWeivel: Type sip, and then press tab |
| 23:18.50 | robin_sz | for some reason my announcements stop after 1, 5 and 7 ... |
| 23:18.56 | Zipper_32 | BullWeivel: Or even just type tab at the open console, |
| 23:19.07 | _VoiceMeUp_COM | Can you pastebin your conf ? |
| 23:19.07 | BSD_Tech | you have to play the first messege |
| 23:19.16 | BSD_Tech | then it will say the rest |
| 23:19.40 | BullWeivel | Zipper_32: when you say console...your talking linux console right?? not like some console within *?? I just did a updatedb and locate sip and i don't see any executeables...yet i have asterisks running. |
| 23:20.10 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 23:20.25 | Zipper_32 | Asterisk console. |
| 23:20.33 | Zipper_32 | asterisk -r |
| 23:20.38 | Zipper_32 | That'll bring you to the asterisk console. |
| 23:20.57 | BSD_Tech | asterisk -vvvvvvvvvvgr |
| 23:20.58 | BullWeivel | okay cool... i was thinking it was an external program :) |
| 23:21.03 | BSD_Tech | is better |
| 23:21.13 | Zipper_32 | BSD_Tech: What's the 'g' for? |
| 23:21.29 | _VoiceMeUp_COM | dump |
| 23:21.33 | _VoiceMeUp_COM | i think ? |
| 23:22.16 | Zipper_32 | "Remove resource limit on core size, thus forcing Asterisk to dump core in the unlikely event of a segmentation fault." Interesting |
| 23:23.50 | _VoiceMeUp_COM | yeah.. |
| 23:24.00 | _VoiceMeUp_COM | guess iw as right.. someone would of jumped on me if i wasnt |
| 23:24.42 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:25.33 | BSD_Tech | lol |
| 23:27.07 | Bananaskin | hey BSD_Tech |
| 23:27.19 | Bananaskin | this SCCP protocol is the dogs bollocks |
| 23:27.46 | BSD_Tech | its the dog balls ? |
| 23:27.49 | BSD_Tech | tmi |
| 23:27.52 | BSD_Tech | lol |
| 23:28.06 | Bananaskin | just grabbed Cisco IP Communicator, using the Hint Facility, BLF etc on the two phones, even shows the number that the other phone has called (on that ext) |
| 23:29.12 | BSD_Tech | brb |
| 23:29.19 | BullWeivel | Dialplan configuration...shouldnt that be in the extensions.conf file and NOT the sip.conf file?? |
| 23:29.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@70.158.103.10) |
| 23:30.08 | BullWeivel | for example: [internal] exten => 100,1,Dial(SIP/john) exten => 611,1,Echo( ) shouldnt that be in the extensions.conf file ? |
| 23:31.07 | [TK]D-Fender | BullWeivel, Yes |
| 23:31.24 | BullWeivel | book is kinda confusing... to me it sounds like from reading it that should go in the sip.conf file. especially since the portion of the book were in is named configuring SIP. |
| 23:31.32 | BullWeivel | Page 71 :) |
| 23:33.49 | Mattwj2005 | Happy Memorials Day for everyone in here that is American |
| 23:34.07 | Zipper_32 | BullWeivel: Your extensions.conf specifies how everything should be routed. sip.conf is only configuration for sip devices. |
| 23:34.14 | Mattwj2005 | :) |
| 23:35.03 | BullWeivel | Zipper_32: well thats what i thought... but when you read the book about setting up SIP.conf it goes on and on..then tells you to try calling yourself by putting in that [internal] info i showed above. it doesnt tell you to go to extensions.conf to do that. |
| 23:35.26 | Zipper_32 | Well that's why we're here to help. =) |
| 23:36.25 | BullWeivel | well thanks for answering :) I know most people hate answering n00b questions. but we were all n00bs at one time ;) |
| 23:36.33 | BullWeivel | brb.. x-lite wants me to reboot. |
| 23:36.35 | *** part/#asterisk BullWeivel (n=bullweiv@12.165.69.254) |
| 23:37.37 | Zipper_32 | Alright, this should be my last issue of the day; Does anybody know how to use an FXS port to connect to a PA system to output sound? I was under the impression that I could just Dial that Zap channel to the PA system, similar to how one can Dial(console/dsp). The Zap channel just rings infinately though. |
| 23:37.42 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:37.42 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:39.36 | tzafrir_laptop | PA system == ? |
| 23:40.10 | Zipper_32 | Public Address. |
| 23:40.13 | tzafrir_laptop | if you call to an FXS port, you need a phone on the other side to pick up |
| 23:40.17 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 23:40.50 | tzafrir_laptop | hopefully the system on the oher side will pick up automatically |
| 23:41.33 | tzafrir_laptop | when the telco calls your phone, you decide if you want to pick up or not. an FXS is the telco here |
| 23:41.34 | Zipper_32 | tzafrir_laptopq: That's what I'm trying to figure out I guess. The PA system has a *tel* port on it with two screws for wiring in the Tip and Ring, |
| 23:42.08 | Zipper_32 | It doesn't seem to be auto answering right now. |
| 23:43.08 | *** join/#asterisk BullWeivel (n=bullweiv@12.165.69.254) |
| 23:43.12 | BullWeivel | im back :) |
| 23:43.19 | BullWeivel | phone don't work..but im back. |
| 23:43.43 | Zipper_32 | BullWeivel: Where are you in configuring right now? |
| 23:44.04 | BullWeivel | im just trying to setup one extension..so it can call itself..im using x-lite |
| 23:44.11 | tzafrir_laptop | Zipper_32, it is an FXO device, right? (not something you connect a phone to) |
| 23:44.16 | BullWeivel | i thought it was registering...but then again... im not seeing it within sip commands. |
| 23:44.42 | BullWeivel | sip show ???? will show me what usres are logged in? |
| 23:44.57 | Zipper_32 | tzafrir_laptop: I believe so. But I do have a x100p I might try interfacing it with. |
| 23:45.12 | Zipper_32 | BullWeivel: sip show peers |
| 23:45.14 | tzafrir_laptop | try a simple phone |
| 23:45.23 | Zipper_32 | Type 'sip show' and then press 'tab' |
| 23:45.49 | Zipper_32 | tzafrir_laptop: I suppose I will. Thanks for the help. |
| 23:45.52 | BullWeivel | yeah i was doing sip show users wondering why it didnt show very much info about the connection :) |
| 23:46.03 | tzafrir_laptop | is it a peer? a user? a friend? |
| 23:46.24 | BullWeivel | im being lazy...its working though. ship show peers shows its ip and everything. |
| 23:46.27 | BullWeivel | its a friend |
| 23:46.47 | Zipper_32 | And it shows the latency? |
| 23:46.49 | BullWeivel | can you dial someone from asterisk and make the phone ring? |
| 23:46.54 | BullWeivel | yeah show latency of like 135s |
| 23:46.59 | tzafrir_laptop | if its IP address is dynamic, 'sip show peers' will show it nicely. |
| 23:47.04 | BullWeivel | ms |
| 23:47.15 | Zipper_32 | BullWeivel: You're almost there. Now all you need to do is make sure you have entries for those SIP devices in the extensions.conf |
| 23:47.28 | Zipper_32 | What is one of your SIP numbers? |
| 23:47.49 | BullWeivel | i thought 100...let me check extensions.conf |
| 23:48.05 | Zipper_32 | No no, your sip show peers will show you your sip numbers |
| 23:48.15 | BullWeivel | exten => 100,1,Dial(SIP/john) |
| 23:48.27 | BullWeivel | oh realy...let me see...dont recall seeing anything in that |
| 23:48.47 | Zipper_32 | what are your sip numbers when you type sip show peers? |
| 23:48.52 | BullWeivel | john/john 10.10.20.200 D 13834 OK (133 ms) |
| 23:48.57 | BullWeivel | no numbers in there. |
| 23:49.03 | BullWeivel | name/username maybe? |
| 23:49.05 | Zipper_32 | oh, well then. Just 'john' |
| 23:49.08 | Zipper_32 | That's what I was inferring. |
| 23:49.23 | Zipper_32 | I've always used extension numbers. ie: 100/100 10.10.20.200 D 13834 OK (133 ms) |
| 23:49.47 | BullWeivel | yeah i was going off the example in the book. |
| 23:49.47 | Zipper_32 | and then in extensions.conf, I would have: exten => 100,1,Dial(SIP/100) |
| 23:50.22 | BullWeivel | whats interesting...if i dial john i get person unavailable please try again. |
| 23:50.36 | Zipper_32 | Give it a try my way, change the sip.conf parts that say 'john' to 100. |
| 23:50.54 | Zipper_32 | And in extensions.conf, change your entry to: exten => 100,1,Dial(SIP/100) |
| 23:51.11 | BullWeivel | brb...let me change all that. |
| 23:51.11 | Zipper_32 | Reload (from the asterisk console) and then try dialing. |
| 23:51.51 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 23:53.55 | BSD_Tech | x100p = crap |
| 23:54.04 | BSD_Tech | and I bet its a clone |
| 23:54.42 | BullWeivel | the only file i need to edit is sip.conf right?? (just making sure...noticed a bunch of good info in users.conf) |
| 23:54.54 | Zipper_32 | BSD_Tech: Yes, but it's useful for a paging interface. |
| 23:55.10 | BSD_Tech | ? |
| 23:55.10 | Zipper_32 | BullWeivel: sip.conf, and then extensions.conf to change 'john' to 100 |
| 23:55.21 | BSD_Tech | I use my sounds card for that |
| 23:55.35 | BSD_Tech | or buy a voip paging unit |
| 23:55.35 | Zipper_32 | BSD_Tech: Into what kind of paging system? |
| 23:55.49 | BSD_Tech | depends |
| 23:55.55 | Zipper_32 | What do you use? |
| 23:56.15 | BSD_Tech | I use my polycom speakers |
| 23:56.32 | Zipper_32 | So you don't use your soundcard for output? |
| 23:56.34 | BSD_Tech | now days |
| 23:56.40 | BSD_Tech | not any more |
| 23:56.59 | Zipper_32 | Well when you did use your soundcard for output, what kind of paging system did you use? |
| 23:57.12 | BSD_Tech | but I use to just fead it from a soundcard into a pa |
| 23:57.25 | BSD_Tech | system wich was just a amp and speakers |
| 23:57.45 | BSD_Tech | I have to look beck in my files what have used in the past |
| 23:58.02 | BSD_Tech | I had to make s conversion cable for |
| 23:58.05 | Zipper_32 | It's okay, don't worry about it. =) |
| 23:58.10 | BSD_Tech | been many years |
| 23:58.35 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 23:59.26 | BullWeivel | 100/100 10.10.20.200 D 40394 OK (1494 ms) |
| 23:59.36 | *** part/#asterisk Mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 23:59.37 | BullWeivel | interesting my ms went way up. |