| 00:00.13 | Linx | just before that you see src= |
| 00:00.20 | Linx | which actually lists the path to the file |
| 00:00.34 | Linx | ls /tmp/ZReqVsAH6qAeMx6ZUrGWDN==.wav |
| 00:00.41 | Linx | . /tmp/ZReqVsAH6qAeMx6ZUrGWDN==.wav |
| 00:00.43 | sandorp | yea, and I'm wondering if the == is part of the file name an causing a parse error |
| 00:00.45 | Linx | so the file exists |
| 00:01.00 | Linx | they have always had == at the end |
| 00:01.05 | Linx | even when it was working |
| 00:01.11 | Linx | even when we used to use mp3 format |
| 00:01.56 | sandorp | hmm, I guess I'm not being much help then :( |
| 00:02.18 | Linx | just my brian is fried and I am not sure where to look now for an answer |
| 00:03.07 | sandorp | don't know but it looks like a problem opening the file and that usually means the file is not there or permissions are too restrictive |
| 00:04.02 | sandorp | and you've checked both so I'm not sure what else it could be ... I'd try dropping the == just for kicks |
| 00:04.34 | _Sam-- | i've 'made install' zaptel1.4.6, but when i try to modprobe, i get not found: root@phone:/usr/src/ast2/asterisk-1.4.14# modprobe zaptel |
| 00:04.34 | _Sam-- | FATAL: Module zaptel not found. |
| 00:04.42 | _Sam-- | any ideas? |
| 00:04.49 | Linx | that unfortunatly is something thats put there by openvxi |
| 00:04.53 | Linx | the vxml browser |
| 00:04.59 | _Sam-- | ive even copied the zaptel.ko where its supposed to go, and same thing |
| 00:05.06 | Linx | not sure how to override what its saved as |
| 00:05.15 | Linx | as its a cache |
| 00:07.25 | Linx | what should the sample rate be on wav files? |
| 00:11.09 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 00:17.35 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 00:19.28 | De_Mon | did I miss something? |
| 00:19.29 | De_Mon | http://pastebin.ca/798759 |
| 00:20.07 | Linx | hmmm seems to be wanting wav49 files |
| 00:20.55 | [hC] | anyone using polycom firmware 2.2.0 here with the new "ring" option instead of "beep" for call waiting? |
| 00:32.30 | *** part/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 00:34.34 | _Sam-- | [TK]D-Fender : lemme know when you get back |
| 00:37.02 | *** join/#asterisk tripppy (n=u@60-242-11-223.static.tpgi.com.au) |
| 00:39.54 | tripppy | is it possible for asterisk to use a serial or PCI dial-up modem with or as a phone? (ATA adapter/PSTN) |
| 00:43.22 | JT | no |
| 00:45.35 | nestAr | it's possible as a FXO |
| 00:45.40 | nestAr | just not as a FXS |
| 00:47.15 | tripppy | so being possible and a FXO, this allows for what? |
| 00:49.55 | Navion | So, I set up paging through the console audio port according to the WiKi but is still seems to not work. Anyone have any experience with this? |
| 00:50.08 | Navion | ~chan_oss |
| 00:51.45 | Navion | No ringing generator in a modem... |
| 00:51.45 | JT | Navion: no, the simple answer is it's NOT possible |
| 00:51.45 | JT | sorry |
| 00:51.45 | JT | nestAr: i meant |
| 00:51.45 | Navion | No way to supply loop voltage either I guess |
| 00:51.45 | JT | nestAr: it's normally NOT possible to use a modem as an FXS or FXO port |
| 00:51.47 | JT | there is one pci modem chipset that has a driver to use it as FXO |
| 00:51.54 | nestAr | yeah, i have one |
| 00:51.59 | nestAr | it works "fair" |
| 00:52.05 | nestAr | haven't used it in a few years. |
| 00:52.05 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:52.22 | JT | nestAr: it's misleading to tell someone that "yes, you can use random modems" |
| 00:52.32 | tripppy | whats the type model? |
| 00:52.37 | Navion | I think that's the one that the O'Relly book says "Ya you can use it but don't..." |
| 00:52.48 | tripppy | lol ok |
| 00:53.11 | tripppy | so i cant use a phone into a modem to call or receive SIP calls? |
| 00:53.24 | nestAr | JT: well, i didn't really want to get into a semantic arguement. But ok. |
| 00:54.36 | JT | nestAr: well you were the one answering his question :) |
| 00:54.37 | nestAr | tripppy: there is a certain motorola chipset that will work as a FXO, they are probably getting harder to find, and they do not work as well as the Digium FXO card, but they are dirt cheap, i paid $9 for mine. |
| 00:54.46 | Navion | No way to ring the phone or supply loop voltage to it with the hardware that a modem contains. Forgetting all together that the firmware is all wrong. |
| 00:56.10 | Navion | Anyone have console audio port paging working? |
| 00:56.26 | tripppy | ok then., would it be possible if i connected my dialup modem to a PSTN network, call out while using a SIP client on the network? |
| 00:57.06 | fujin | ~cheap |
| 00:57.06 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 00:57.17 | JT | tripppy: no, just buy an ATA or PCI telephony card. |
| 00:57.21 | fujin | ^^. |
| 00:57.51 | tripppy | yeah ok. just exploring my options i have, i own 10xdialup modems |
| 00:58.08 | JT | yeah, best to forget about them :) |
| 00:58.21 | [hC] | I dont suppose there is an option in asterisk 1.4 to timestamp each line on the CLI output? |
| 00:58.26 | fujin | chuck them out |
| 00:58.33 | fujin | [hC]: not as far as I know, you can log to a file with timestamps thouhg |
| 00:58.40 | fujin | then obviously you don't get ANSI colour |
| 00:58.55 | fujin | [Nov 27 13:58:47] VERBOSE[10817] logger.c: == Spawn extension (macro-delivercall, sw-54-BUSY, 10) exited non-zero on 'Local/735@agents-651d,2' |
| 00:59.50 | fujin | hrm |
| 00:59.55 | fujin | my CDR to mysql has mysteriously broken |
| 01:00.36 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 01:00.45 | Linx | what should the bitrate and sample rate be for wav pcm files for asterisk? |
| 01:00.53 | UserReg_CL | Hi !!! Helpme please :) |
| 01:01.14 | Qwell | ~help |
| 01:01.19 | Nivex | ~data |
| 01:01.20 | jbot | Don't Ask To Ask. Just ASK |
| 01:01.21 | Qwell | stupid bot |
| 01:01.25 | Nivex | jbot: wake up! |
| 01:01.26 | jbot | ACTION throws a barrel-full of ice water on up! and shouts "GOOD MORNING!!!!" |
| 01:01.29 | UserReg_CL | ~lol |
| 01:01.30 | jbot | rumour has it, lol is stands for Laughing Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
| 01:01.35 | Qwell | <LinuxHOW2> help is probably best gained by asking specific questions and providing ALL relevant details about the distribution being used and the software involved and ALL pertinent errors messages! |
| 01:01.48 | fujin | what do I need to do to reload cdr_addon_mysql? I can't see it in reload ? |
| 01:01.52 | UserReg_CL | what distro linux your recomend for install asterisk ? |
| 01:01.55 | fujin | [Nov 27 14:01:46] NOTICE[10702]: cdr.c:434 ast_cdr_free: CDR on channel 'Local/735@agents-75ea,1' not posted |
| 01:02.03 | Qwell | UserReg_CL: whatever you're familiar with |
| 01:02.07 | grimsy | UserReg_CL: fedora/redhat/centos |
| 01:02.15 | fujin | ew |
| 01:02.17 | Qwell | UserReg_CL: most of the developers use Debian |
| 01:02.18 | grimsy | but yeah, what Qwell said |
| 01:02.18 | fujin | that's disgusting |
| 01:02.24 | fujin | please don't ever suggest a RH derivative ever again |
| 01:02.24 | Qwell | or, a debian of some type |
| 01:02.35 | Qwell | fujin: I quite like suse |
| 01:02.48 | grimsy | i thought asterisk was originally developed on a rh system? |
| 01:02.57 | UserReg_CL | Qwell what version debian ? |
| 01:03.02 | Qwell | grimsy: rh was relevant 10 years ago :p |
| 01:03.05 | fujin | trixbox/a@h was developed on Centos, afaik |
| 01:03.08 | grimsy | :D |
| 01:03.16 | UserReg_CL | CentOS ? |
| 01:03.20 | UserReg_CL | ~centos |
| 01:03.21 | jbot | centos is a rebuild of the Red Hat Enterprise Linux RPMs by the community. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 01:03.37 | Qwell | heh |
| 01:03.43 | fujin | use debian/ubuntu |
| 01:03.48 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 01:04.27 | UserReg_CL | Qwell: your know about AGI ? |
| 01:04.28 | Qwell | jbot: no, centos is a rebuild of a predominant North American Linux vendors RPMs, by the community. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 01:04.29 | jbot | okay, Qwell |
| 01:04.41 | grimsy | haha |
| 01:04.57 | Qwell | jbot: no, centos is a rebuild of a prominent North American Linux vendors RPMs, by the community. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 01:04.58 | jbot | okay, Qwell |
| 01:05.00 | Qwell | typo |
| 01:05.05 | JT | jbot.,. centos also comes qith free spinlock kernel bugs |
| 01:05.11 | JT | with |
| 01:05.19 | Qwell | jbot: no, centos is a rebuild of a prominent North American Enterprise Linux vendor's RPMs, by the community. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 01:05.20 | jbot | okay, Qwell |
| 01:05.30 | Qwell | there |
| 01:05.33 | UserReg_CL | have one problem with one very small agi |
| 01:05.43 | Qwell | screw it, I might as well go marketing |
| 01:05.57 | Qwell | jbot: no, CentOS is an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. |
| 01:05.57 | jbot | okay, Qwell |
| 01:06.04 | Qwell | straight from centos.org :p |
| 01:06.20 | Qwell | jbot: no, CentOS is an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 01:06.20 | jbot | Qwell: okay |
| 01:06.32 | Qwell | </too much fun> |
| 01:06.48 | JT | centos is a repackage of an "enterprise" linux distro without any of the support that enterprises want |
| 01:06.59 | JT | in other word, .rpm pain for no gain |
| 01:07.01 | Qwell | They're very careful to not say redhat |
| 01:07.04 | _Sam-- | after successful 'make install' of zaptel 1.4.6, and after i rmmod zaptel, i am unable to load new module...root@phone:/var/log/asterisk# modprobe zaptel |
| 01:07.04 | _Sam-- | FATAL: Module zaptel not found. |
| 01:07.08 | UserReg_CL | ~agi |
| 01:07.09 | jbot | somebody said agi was the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 01:07.09 | fujin | JT: well played |
| 01:07.24 | _Sam-- | any ideas? |
| 01:07.34 | Qwell | JT: well, on the other side of things, you've got Ubuntu |
| 01:07.39 | fujin | _Sam--: it's not in the right place |
| 01:07.43 | UserReg_CL | _Sam: use debian or other distro ? |
| 01:07.45 | _Sam-- | ive put it everywehere |
| 01:07.50 | _Sam-- | debian (old sarge) |
| 01:07.58 | fujin | You shouldn't be putting it anywhere by hand |
| 01:08.03 | _Sam-- | i put the .ko's manually in the same place as my last ones, just to try. |
| 01:08.10 | fujin | o_0 |
| 01:08.15 | fujin | can you 'insmod' the .ko? |
| 01:08.23 | _Sam-- | lemme try that |
| 01:08.36 | _Sam-- | root@phone:/var/log/asterisk# insmod zaptel |
| 01:08.36 | _Sam-- | insmod: can't read 'zaptel': No such file or directory |
| 01:08.39 | fujin | nub |
| 01:08.42 | fujin | man insmod |
| 01:08.54 | fujin | why would you put 'zaptel.ko' in /var/log/asterisk |
| 01:09.02 | fujin | and more importantly, why would you try and use insmod like modprobe |
| 01:09.12 | fujin | modprobe relies on a local cache of where modules are |
| 01:09.19 | _Sam-- | insmod worked. |
| 01:09.20 | fujin | insmod inserts .ko's directly |
| 01:09.26 | fujin | cool |
| 01:09.29 | _Sam-- | i didn tput it in /var/log/asterisk, either |
| 01:09.31 | fujin | so update your module cache |
| 01:09.32 | _Sam-- | that was just the wd |
| 01:09.39 | Qwell | run umm...what's that command? |
| 01:09.44 | fujin | pass |
| 01:09.46 | fujin | I forget |
| 01:09.51 | fujin | depmod -a |
| 01:09.55 | _Sam-- | thanks! |
| 01:09.57 | Qwell | yeah |
| 01:10.05 | Qwell | I can never remember that one |
| 01:10.11 | Qwell | it should be moddep, imo |
| 01:10.17 | fujin | lol, yeah |
| 01:10.21 | fujin | at least then you could modtap |
| 01:10.22 | fujin | tab |
| 01:10.24 | Qwell | I always try to mod<tab> |
| 01:10.24 | fujin | <tab> |
| 01:10.24 | Qwell | yeah |
| 01:10.33 | fujin | ^5 Qwell |
| 01:10.40 | Qwell | I always think "modules-update?" |
| 01:10.47 | fujin | That's for Gentoo, no? |
| 01:10.57 | Qwell | yes |
| 01:11.01 | fujin | yeah. thought so. |
| 01:11.03 | fujin | Gentoo > * |
| 01:11.13 | JT | Qwell: what about ubuntu? |
| 01:11.18 | _Sam-- | thanks, both. |
| 01:11.26 | fujin | Ubuntu is great |
| 01:11.32 | Qwell | JT: repackaged debs for the sake of repackaging |
| 01:11.44 | UserReg_CL | Helpme AGI script: http://pastebin.com/m2d6f18be .. Thank |
| 01:11.47 | Qwell | .deb pain for no gain |
| 01:11.54 | JT | Qwell: no. |
| 01:11.57 | fujin | I find ubu to have less pain that debian sometimes |
| 01:11.57 | JT | it's not a repackage |
| 01:11.59 | UserReg_CL | is very easy |
| 01:12.02 | fujin | s/that/than/ |
| 01:12.05 | Qwell | much of it is |
| 01:12.09 | JT | ubuntu is a deb based distro |
| 01:12.21 | fujin | JT: alot of it is based around upstream |
| 01:12.27 | JT | they have put a lot of groundwork into user friendlyness |
| 01:12.28 | JT | sure |
| 01:12.35 | fujin | but yes, alot of work is done in the ubu repositories |
| 01:12.38 | Qwell | granted, there are a lot more changes than centos |
| 01:12.39 | JT | but the aim is not to copy a distro and relicense it |
| 01:12.49 | Qwell | same basic premise though :p |
| 01:12.57 | JT | eh sure, it's linux |
| 01:13.06 | Qwell | something like gentoo, however... |
| 01:13.08 | UserReg_CL | :( |
| 01:13.18 | fujin | GENTOO WINS~ |
| 01:13.18 | Qwell | completely rethought from the ground up |
| 01:13.22 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 01:13.39 | fujin | Qwell: if rethought coutns as 'stole some ideas from bsd' |
| 01:13.39 | fujin | then sure |
| 01:13.50 | Qwell | the only thing I wish gentoo had, was official binary builds being available for common USE flags |
| 01:13.53 | fujin | but I must say, the bsd-style init system of Gentoo is awesome |
| 01:13.59 | fujin | Qwell: yeah, that'd be handy |
| 01:14.03 | fujin | time taken vs. time spent |
| 01:14.19 | UserReg_CL | helpme: agi script no work |
| 01:22.04 | UserReg_CL | mmm |
| 01:23.24 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:23.40 | obnauticus | Is it possible to plug the pair of my headset port into my computer and use err |
| 01:23.50 | obnauticus | my sound card and window's settings on it |
| 01:28.29 | *** join/#asterisk radicall (n=Zynth@189.168.192.6) |
| 01:32.40 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:37.09 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-6cfc6cfa3d8989e6) |
| 01:37.51 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:40.46 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 01:44.24 | UserReg_CL | mmm Thank !!! |
| 01:44.31 | UserReg_CL | for all ... good bye |
| 01:48.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:49.43 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
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| 02:14.15 | dijungal | hi, i have polycom 430s that keeps restarting on me randomly |
| 02:14.19 | dijungal | any reason why? |
| 02:16.10 | Qb3rt | Hi, iam working on an asterisk project for a bank and they want to record every asterisk conversation on a encrypted drive... Is that possible??? How do i encrypt a drive in linux? Is asterisk will be able to write on that encrypted drive?? |
| 02:17.07 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 02:19.05 | *** join/#asterisk simprix (n=dschuema@c-71-205-52-252.hsd1.mi.comcast.net) |
| 02:19.17 | simprix | How can I set the tos to 5 in asterisk 1.2 |
| 02:21.42 | dijungal | simprix: ??? huh?? |
| 02:21.59 | dijungal | simprix: tos = |
| 02:23.17 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 02:23.45 | dijungal | hi, i have polycom 430s that keeps restarting on me randomly. Should I update to the latest firmware? |
| 02:26.15 | *** join/#asterisk Odie_flocon (n=odie@S0106000f3d5eae81.cg.shawcable.net) |
| 02:27.51 | Odie_flocon | hello all. |
| 02:29.40 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 02:29.57 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 02:30.33 | znoG | hey all |
| 02:31.34 | *** join/#asterisk salviadud (i=ralfalfa@voss.dreamhost.com) |
| 02:32.15 | znoG | in my dialplan, I'd like to allow all SIP/IAX extensions to dial out, only if they're coming from the local network. If the person connects to the Asterisk box remotely, then I don't want to allow outgoing local calls. I thought the way to go would be using AGI and looking at some of the channel variables to determine the IP of the incoming request ... am I on the right track? |
| 02:32.47 | JT | err |
| 02:32.55 | JT | why not use a user account for each extension? |
| 02:34.14 | znoG | I have a user account for each extension, but I rather use the same account whether I'm connecting locally or remotely .. hence the AGI idea |
| 02:34.54 | znoG | you're suggesting to use 2 user accounts per extensions and permit/allow statements, etc? |
| 02:35.37 | _ShrikE | znoG look at function sippeer |
| 02:35.52 | _ShrikE | pull the ip and see if its on your internal network |
| 02:36.48 | znoG | what about for IAX? is there something similar? |
| 02:37.06 | _ShrikE | function iaxpeer |
| 02:37.37 | znoG | that was unexpected ;) |
| 02:38.00 | znoG | so to find out if its in my internal network, should I just use some php/perl/python/etc AGI script? |
| 02:38.10 | JT | 2 seperate accounts would be ideal |
| 02:38.21 | JT | agi seems like excessive complication |
| 02:38.42 | _ShrikE | Or just gotoif on the results if sip/iaxpeer ip. |
| 02:38.46 | *** join/#asterisk etfonhomey (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
| 02:38.48 | znoG | JT: yea, the thing is I already have 2 user accounts per extension (SIP/IAX) .. so I'd have to add a third account for connecting from outside .. |
| 02:39.54 | *** join/#asterisk usam (n=aya@202.91.19.194) |
| 02:40.16 | JT | sip and iax don't have to have different account |
| 02:40.23 | JT | don't usually have to have iax either ;) |
| 02:40.52 | usam | Hello, i got an request from my consultee that an carrier will send their traffic via TDM, can asterisk handle this? I want to do iax trunking from 2 points |
| 02:41.21 | usam | between 2 points i mean . . |
| 02:41.30 | JT | usam: you're not making much sense |
| 02:41.32 | JT | it's either TDM |
| 02:41.35 | JT | or IAX |
| 02:41.58 | znoG | JT: they don't? the idea is to use SIP internally and IAX when connecting from remote |
| 02:42.14 | znoG | due to firewall restrictions, we wanted to use IAX to connect from remote |
| 02:42.40 | znoG | but I don't understand what you mean by 'sip anx iax don't have to have different account' |
| 02:42.44 | JT | well if that's the case you already have 2 accounts |
| 02:42.52 | JT | realtime or users.conf |
| 02:42.53 | znoG | true, i was thinking that as I typed |
| 02:43.06 | usam | JT: lets sat carrier A send traffic til me via tdm, i trans-protocol to IAX to another point and trans-protocol tol sip |
| 02:43.10 | znoG | yes, so IAX will be used remotely .. and SIP locally |
| 02:43.13 | JT | why can't you allow sip through the firewall? |
| 02:43.15 | znoG | SIP I can restrict internally |
| 02:43.21 | usam | s/sat/say |
| 02:43.36 | JT | usam: asterisk can connect to PRI |
| 02:43.42 | znoG | JT: well the asterisk box is behind NAT .. and the clients will most likely be behind NAT too .. which, as I understand it, causes firewall hell |
| 02:43.51 | JT | znoG: not if setup correctly. |
| 02:43.53 | JT | ~sipnat |
| 02:43.54 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:44.07 | usam | JT: ok lemme chekc out the PRI stuff then |
| 02:44.16 | usam | thx for the info |
| 02:44.26 | JT | you portforward sip and rtp to the asterisk box, set externip= and localnet= and nat=yes |
| 02:44.46 | JT | and use qualify=yes and make sure clients register, and they should usually work |
| 02:45.05 | JT | no port forwarding on the client end |
| 02:50.02 | znoG | and the client can be behind NAT without any firewall mods? |
| 02:50.14 | JT | right |
| 02:50.20 | znoG | interesting .. |
| 02:50.31 | JT | well obviously there can't be deny rules blocking the traffic |
| 02:50.36 | znoG | right |
| 02:50.42 | JT | but this is how hundreds of ITSPs operate |
| 02:50.54 | JT | the ATAs they give customers speak SIP not IAX |
| 02:51.02 | znoG | so basically i could bring it down to 1 SIP account .. with some GotoIf magic and SIPPEER to determine where the connection came from |
| 02:51.23 | znoG | so that I can allow/deny outbound local calls |
| 02:55.17 | *** part/#asterisk simprix (n=dschuema@c-71-205-52-252.hsd1.mi.comcast.net) |
| 02:56.54 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 03:00.21 | BlackSlik | II have this on my sip.conf |
| 03:00.22 | BlackSlik | Note: If your SIP devices are behind a NAT and your Asterisk |
| 03:00.22 | BlackSlik | ; server isn't, try adding "nat=1" to each peer definition to |
| 03:00.22 | BlackSlik | ; solve translation problems. |
| 03:00.22 | BlackSlik | [general] |
| 03:00.23 | BlackSlik | bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) |
| 03:00.25 | BlackSlik | bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) |
| 03:00.35 | BlackSlik | bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) |
| 03:00.36 | BlackSlik | disallow=all |
| 03:00.36 | BlackSlik | allow=ulaw |
| 03:00.36 | BlackSlik | allow=alaw |
| 03:00.36 | BlackSlik | context = from-sip-external ; Send unknown SIP callers to this context |
| 03:00.37 | BlackSlik | callerid = Unknown |
| 03:00.39 | BlackSlik | #include sip_nat.conf |
| 03:00.41 | BlackSlik | #include sip_custom.conf |
| 03:00.43 | BlackSlik | #include sip_additional.conf |
| 03:00.45 | BlackSlik | #include additional_a2billing_sip.conf |
| 03:00.47 | BlackSlik | where do i add my sip account details |
| 03:02.11 | JT | BlackSlik: STOP |
| 03:02.16 | JT | stop flooding us |
| 03:02.24 | JT | ~pb |
| 03:02.25 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:03.01 | JT | BlackSlik: trixbox/freepbx is not supported here |
| 03:03.02 | JT | ~tribox |
| 03:03.08 | JT | ~trixbox |
| 03:03.09 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 03:03.34 | znoG | JT: am I right with what I'm saying? GotoIf magic and SIPPEER to determine if the outbound call request should be allowed or not? |
| 03:04.16 | JT | you could, seems like a complete hack though |
| 03:04.27 | JT | much better for extensions to be determined via context |
| 03:06.39 | znoG | so one SIP user with permit/deny (internal) and another SIP user with a more restricted context? |
| 03:10.31 | JT | right |
| 03:10.39 | JT | don't forget you can include contexts |
| 03:12.38 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 03:12.58 | *** join/#asterisk PepOSX (n=pepOSX@190.72.147.37) |
| 03:13.20 | *** join/#asterisk the007killer (n=the007ki@61.29.2.98) |
| 03:13.37 | the007killer | hi everyone |
| 03:13.58 | *** join/#asterisk gardo (n=gardo@121.97.108.105) |
| 03:15.26 | BlackSlik | JT sorry for the long paste, Actually this is what i am trying to archive, i have want to install asterisk and Be able to add a sip account to it (Mean while i have no sip device So i want it 2 be kind of vitrual) , and be able to make 10000 outbond calls from the asterisk box with a wav which calls the pstn number at same time |
| 03:16.13 | the007killer | does anyone here have experience with compiling mysql with asterisk addons? |
| 03:16.57 | JunK-Y | the007killer: just make sure you have mysql lib and there you go for compiling ur addon. |
| 03:21.19 | JT | BlackSlik: then delete and reinstall asterisk without trixbox/freepbx |
| 03:21.25 | JT | and take a look at ~thebook |
| 03:21.57 | JT | and i doubt you'll have a provider that can send 10000 outbound calls at once, let alone machine(s) powerful enough |
| 03:22.01 | JT | ~thebook |
| 03:22.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 03:22.39 | BlackSlik | JT i have a voip account to handle such calls |
| 03:25.36 | the007killer | it seemd to break at the configure part of the asterisk addons |
| 03:25.37 | the007killer | checking for mysql_config... /usr/bin/mysql_config |
| 03:25.37 | the007killer | checking for mysql_init in -lmysqlclient... no |
| 03:25.52 | BlackSlik | JT are there anyother tools to use in configuring asterisk on an explorer |
| 03:26.35 | JunK-Y | the007killer: you need to install mysql lib. |
| 03:27.28 | the007killer | how do i install that? (im not familiar with centos) |
| 03:27.38 | JT | BlackSlik: how much outbound bandwidth do you have? |
| 03:30.38 | the007killer | woooooooooooooooo it works |
| 03:30.49 | the007killer | sorry been trying to get this to work for 2 weeks now |
| 03:34.25 | BlackSlik | JT is about 500Mb |
| 03:34.29 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 03:34.33 | *** part/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 03:35.51 | JT | BlackSlik: be more specfic, 500Mbit/s or 500MByte/s? |
| 03:36.02 | JT | BlackSlik: and what cluster do you have to send the calls? |
| 03:37.57 | BlackSlik | JT i am not to technical on the asterisk, all i am lookin at is archiving that either i get someone 2 handle it or i get some who i can pay to tell me step by step |
| 03:38.48 | JT | BlackSlik: it is a massive engineering project |
| 03:38.58 | JT | 10000CPS is at the telco level |
| 03:39.11 | JT | and probably an unreasonable request that your customer has made |
| 03:39.30 | JT | why do they need so many concurrent calls? |
| 03:39.37 | BlackSlik | well not 10000 at same time maybe like 100calls to be made like voice broadcasting |
| 03:39.56 | BlackSlik | in a particular time frame |
| 03:39.58 | JT | how many concurrent calls, how many new calls per second? |
| 03:40.09 | BlackSlik | 100 |
| 03:40.28 | JT | they are different numbers |
| 03:40.33 | JT | which question did you answer? |
| 03:41.01 | BlackSlik | i answered the concurrent calls |
| 03:41.13 | BlackSlik | they are different pstn american numbers |
| 03:41.15 | JT | and CPS? |
| 03:41.50 | BlackSlik | CPS means |
| 03:42.06 | outtolunc | calls per second |
| 03:42.27 | *** join/#asterisk usam (n=aya@202.91.19.194) |
| 03:42.37 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 03:42.41 | JT | i already mentioned it |
| 03:42.47 | JT | but yes, calls per seconds |
| 03:42.51 | JT | -s |
| 03:43.03 | BlackSlik | ok |
| 03:43.40 | BlackSlik | 1000 |
| 03:43.48 | JT | that sir, is illogical |
| 03:44.01 | JT | you will have 100 concurrent calls, but 1000 CPS? |
| 03:44.15 | JT | that means the average call length is 0.1 second |
| 03:44.19 | BlackSlik | yes 100 |
| 03:44.22 | etfonhomey | I'm enjoying this conversation... |
| 03:45.03 | JT | concurrent calls is usually > CPS |
| 03:45.21 | BlackSlik | 100 concurrent calls would be made out with a wav file to each pstn numbers |
| 03:45.41 | JT | concurrent calls means number of channels used |
| 03:45.49 | JT | at an instant |
| 03:45.53 | BlackSlik | the wav file size is about 850kbs |
| 03:46.04 | JT | CPS means the amount of new calls commissioned in a seconfd |
| 03:46.52 | BlackSlik | yesi do understand sir |
| 03:47.31 | JT | if your calls have any real material length i suggest you check your figures of 100 concurrent and 1000 CPS |
| 03:48.10 | BlackSlik | okay, what about the script that makes this calls which my major problem |
| 03:48.35 | JT | the problem is that the figures you gave made no sense |
| 03:48.41 | *** part/#asterisk Linx (n=linx@219-89-195-91.adsl.xtra.co.nz) |
| 03:49.16 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 03:49.28 | BlackSlik | like what figure u think could be a perfect ideal |
| 03:49.42 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 03:50.38 | JT | well the CPS rate needs to be less than the amount of channels used, and exactly how much less depends on the length of calls |
| 03:50.49 | JT | i thought you knew what the requirement was |
| 03:53.14 | BlackSlik | the length of the call is about to be made is about 49 seconds |
| 03:55.56 | *** part/#asterisk dijungal (n=kdaniel@208.0.231.85) |
| 03:56.03 | fujin | uh |
| 03:56.05 | fujin | hey, quick question |
| 03:56.16 | fujin | I'm using a crontab to pull a dial number out of a db, and goign to store it in a file |
| 03:56.24 | fujin | what's thd best way to pull that into a $VARIABLE for asterisk? |
| 03:56.49 | JT | i don't understand the question |
| 03:57.13 | fujin | file "blah" contents are "1021921741" |
| 03:57.22 | fujin | I want 1021921741 into ${dialout} |
| 03:57.32 | outtolunc | 'store it in a file' assumes a 'call file', if so, read sample.call (Variable:) |
| 03:57.40 | fujin | no, not a call file |
| 03:58.10 | fujin | just a flat text file so that if the db is down, calls will still work |
| 03:58.16 | fujin | calls to this variable anyway |
| 03:59.03 | fujin | any ideas? |
| 03:59.05 | outtolunc | that is 2 different questions |
| 03:59.22 | DarkRift | I'd put it as a global variable in the extension rather than reading a file for 1 line |
| 03:59.37 | outtolunc | if you just want to write info you gleened from a db to a flatfile, that has nothing to do with asterisk |
| 03:59.38 | fujin | That's also not what I want. |
| 03:59.44 | fujin | Yes, it does. |
| 03:59.49 | outtolunc | haha |
| 03:59.50 | fujin | I want asterisk to be able to read the file which contains the information |
| 04:00.06 | fujin | I guess I can use func system, but that's dirty. |
| 04:00.23 | outtolunc | why not just use the asterisk DBget/put support? |
| 04:00.29 | fujin | That doesn't do what I want either. |
| 04:00.32 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 04:00.32 | JT | err |
| 04:00.39 | JT | fujin: use csv |
| 04:00.39 | fujin | Jesus, way to complicate what i'm trying to do. Read a string from a text file into a variable |
| 04:00.51 | fujin | JT: it's just a single string |
| 04:01.09 | fujin | comma seperating a single value doesn't seem right |
| 04:01.41 | JT | System maybe |
| 04:01.43 | JT | shrug |
| 04:01.44 | JT | agi? |
| 04:01.48 | fujin | yuck |
| 04:01.52 | fujin | I guess system will do it cleanly enough. |
| 04:02.42 | fujin | actually |
| 04:02.42 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:02.49 | fujin | can I set the output of system to a variable |
| 04:03.08 | DarkRift | I was looking at it |
| 04:03.09 | DarkRift | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System |
| 04:03.11 | DarkRift | Example 4 |
| 04:03.20 | DarkRift | After example 4 rather |
| 04:04.01 | fujin | What, security? |
| 04:04.08 | DarkRift | The example under that |
| 04:04.16 | DarkRift | exten => 200,1,Set(CALLERID(NAME)=PG&/bin/echo BADIDEA > /ROOTED.txt) |
| 04:04.39 | fujin | Yes, and? |
| 04:04.42 | fujin | That doesn't do what I need. |
| 04:04.54 | fujin | I need to read the contents of a file as a string into a ${variable} |
| 04:05.05 | fujin | not set a variable to attempt to exploit System() |
| 04:07.23 | DarkRift | Well as I don't know much on if you can store the content of System to a var, looks like agi doesn't look like a bad idea or ODBC text driver might be able to do it (but that one I have no idea how to make read calls for it) |
| 04:08.00 | fujin | Hrmph |
| 04:08.32 | fujin | Nevermind, I'll use my script to generate some content to #include and then 'ael reload' |
| 04:20.14 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
| 04:21.45 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-250-100-205.dsl.ipltin.ameritech.net) |
| 04:22.27 | perf3kt | do the hardcore asterisk guy recommend the digium training? |
| 04:23.55 | fujin | Ah. The joys of bash. |
| 04:23.58 | fujin | perf3kt: no |
| 04:24.18 | grimsy | fujin: he said hardcore ;) |
| 04:24.48 | fujin | I'm pretty hardcore. |
| 04:25.09 | grimsy | the hardest |
| 04:25.32 | perf3kt | so, the alternative |
| 04:25.33 | fujin | Up there, anyway. |
| 04:25.44 | perf3kt | I mean I wanna get into reselling |
| 04:25.44 | fujin | perf3kt: read the documentation that accompanies the source tarball, idle here |
| 04:25.50 | perf3kt | and eventaully support |
| 04:25.55 | fujin | learn2asterisk then |
| 04:27.10 | perf3kt | wow |
| 04:27.16 | perf3kt | you guys are tons of help |
| 04:27.32 | Nugget | the documentation is even more helpful. |
| 04:27.44 | JT | ~thebook |
| 04:27.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 04:27.50 | perf3kt | i got the book |
| 04:27.54 | perf3kt | I've read the book |
| 04:27.59 | fujin | You haven't stumbled upon a gold-mine of telling you how to make money |
| 04:27.59 | perf3kt | thanks though |
| 04:28.00 | fujin | fwiw |
| 04:28.13 | fujin | learn2asterisk , get good at it, sell it |
| 04:28.17 | fujin | reselling what by the way? |
| 04:28.23 | perf3kt | well I'll tell ya like this |
| 04:28.27 | fujin | reselling something you got for free? (doesn't that just make it selling?) |
| 04:28.33 | perf3kt | not everyone is hardcore, linux |
| 04:28.40 | fujin | Then they're doing it wrong. |
| 04:29.04 | perf3kt | ... |
| 04:29.09 | *** join/#asterisk usam (n=aya@202.91.18.194) |
| 04:29.14 | perf3kt | guess i'm wasting my breath here |
| 04:29.30 | fujin | Probably. |
| 04:29.34 | fujin | try #trixbox |
| 04:29.36 | fujin | or uh |
| 04:29.37 | fujin | that other one |
| 04:29.39 | fujin | callweaver? |
| 04:29.59 | perf3kt | I know them all I wouldn't come here if I didnt' want to ask the asterisk comman line guys |
| 04:30.11 | fujin | so, what's the problem? |
| 04:30.14 | fujin | you're asking the wrong questions? |
| 04:30.20 | fujin | asterisk training = build a system |
| 04:30.35 | fujin | Good day. I'm off home. |
| 04:31.39 | perf3kt | is the new edition of the book out? |
| 04:31.46 | JT | yeah |
| 04:31.54 | [TK]D-Fender | perf3kt, that is the 2nd edition |
| 04:32.01 | alephcom | fujin: that's the easy part. The tough stuff starts when something breaks badly |
| 04:32.13 | perf3kt | ~book |
| 04:32.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 04:35.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:35.45 | perf3kt | I think you all just hate n00bs |
| 04:35.50 | perf3kt | lol |
| 04:36.04 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 04:39.52 | [TK]D-Fender | perf3kt, You just need to put a hold on this line of questioning, and just go and get your hands dirty with Asterisk. |
| 04:40.20 | perf3kt | well my hands have been dirty, honestly |
| 04:40.21 | [TK]D-Fender | perf3kt, We promise that by the time certain individuals around here are through with you, you'll never feel "clean" again ;) |
| 04:40.43 | perf3kt | it jsut is frustrating to run into hurdles of learngin abotu the dependencies, and builds and linux to get to astersisk |
| 04:41.02 | perf3kt | not to mention distro's |
| 04:41.04 | [TK]D-Fender | perf3kt, ..... thats layed out in the first chapter of the BOOK :) |
| 04:41.18 | [TK]D-Fender | perf3kt, and falls rather firmly under the category of "oh please!" |
| 04:41.34 | perf3kt | before ever getting to a asterisk build, well it may be better in the 2nd, but it wasn't in the 1st |
| 04:41.40 | [TK]D-Fender | http://www.asterisk.org/support/install |
| 04:42.10 | [TK]D-Fender | now that link was for 1.2, but its not terribly different. |
| 04:42.39 | Nugget | seems like a nutty complaint, to be honest. as far as software goes, asterisk has very few dependencies. |
| 04:42.53 | [TK]D-Fender | AND its all in the book so anyone who is still "wondering", quite frankly hasn't gotten off their ass to profit from the free distributed works of others :) |
| 04:43.07 | perf3kt | well everyone isn't use to linux and dependenceis and yums and tarballs |
| 04:43.14 | perf3kt | but hey guess that is the nature of the beast |
| 04:43.23 | perf3kt | not complaining just was asking about training that is all |
| 04:43.26 | [TK]D-Fender | Nugget, and has strangly brought us such a wealth of co-dependant people ;) |
| 04:43.37 | perf3kt | I think that is a legtimate request being not very knoledgable and all |
| 04:43.38 | Nugget | if you had trouble with "./configure && make" then what exactly do you expect will be your "V" in your hypothetical "VAR" selling asterisk? |
| 04:43.48 | [TK]D-Fender | perf3kt, If you don't know linux at all well... sorry, there is no mercy for you... |
| 04:44.14 | [TK]D-Fender | Nugget, Vallium! |
| 04:44.25 | Nugget | "R"s don't do very well in the marketplace. The "VA" part is pretty crucial. |
| 04:44.56 | perf3kt | thanks |
| 04:44.57 | [TK]D-Fender | Nugget, VA implies deeper telecom & linux knowledge for real specialization. |
| 04:45.41 | Nugget | Look, nobody denies that asterisk can be confusing and complicated and nearly all of us are in here because at some point we ran into a roadblock we couldn't sort out on our own. |
| 04:45.51 | CunningPike | perf3kt: Clearly, you have never built something like sguil from source, if you think asterisk is hard |
| 04:46.00 | Nugget | are you SERIOUSLY telling us that for you that roadblock was compiling the freakin' program? |
| 04:46.07 | Nugget | that's documented about twenty ways to sunday |
| 04:46.23 | Nugget | on dozens of websites and in a really great book |
| 04:46.40 | perf3kt | thanks |
| 04:46.57 | perf3kt | i'll just got get the book and crawl into a corner |
| 04:46.57 | Nugget | the only logical explanation is that you've spent no actual effort to get it working, so why the heck should we (helping you get started)? |
| 04:47.12 | perf3kt | that isn't logical |
| 04:47.17 | Nugget | of course it is. |
| 04:47.25 | Nugget | either that or you're a total moron, which is unlikely. |
| 04:47.32 | Nugget | since you're articulate and managed to get an irc client working |
| 04:47.42 | Nugget | so we're left with "lazy" as the probably explanation |
| 04:47.49 | Nugget | er, "probable" |
| 04:47.56 | perf3kt | no, I'll tell you the situation |
| 04:48.05 | perf3kt | about 6monyths ago I started into this |
| 04:48.11 | perf3kt | had the first edition |
| 04:48.14 | perf3kt | did alot of reading |
| 04:48.18 | perf3kt | started looking online |
| 04:48.34 | perf3kt | started into the base asterisk from source, got it going |
| 04:49.02 | perf3kt | then I saw that the gui based did all of that in one cd, one step, I said why am I busting my balls |
| 04:49.25 | perf3kt | spending countless hours on yums that weren't working |
| 04:49.29 | perf3kt | and etc |
| 04:49.47 | [TK]D-Fender | perf3kt, because those cookie-cutter distros make you do things in its tiny limited little way. |
| 04:49.49 | perf3kt | then I went to those and got somethings working, calls, ivrs, etc |
| 04:49.51 | [TK]D-Fender | perf3kt, thats why |
| 04:49.56 | perf3kt | hold |
| 04:50.10 | perf3kt | but realizing that I didn't have a full understanding I came back here |
| 04:50.15 | [TK]D-Fender | perf3kt, If any chump can have their "miracle" setup with just a few clicks... who needs **YOU**? |
| 04:50.26 | [TK]D-Fender | perf3kt, your "value" has just EVAPORATED |
| 04:50.32 | perf3kt | to ask for a definitive way from command to do it |
| 04:51.06 | perf3kt | well its being sold at trixbox, at elastix |
| 04:51.39 | [TK]D-Fender | perf3kt, yes... but who needs YOU then? These are GUI'd dead-ends that have nothing to do with your skill and are locked to their way of thinking. |
| 04:51.44 | perf3kt | and they've even bought a couple of your command line buddies to make it work and support it |
| 04:52.26 | [TK]D-Fender | perf3kt, Those that just want a cookie-cutter setup, sure, go enjoy Trixbox/etc. For those who want real control and to do "intersting" stuff... THOSE are the ones that will learn to roll their own. |
| 04:52.43 | perf3kt | honestly that is what I'm trying to figure out, what is the difference, what is the value of the command line vs the gui |
| 04:53.24 | perf3kt | i mean its no secret that the gui can be sold |
| 04:53.42 | perf3kt | companies and sellers find value in the gui |
| 04:53.56 | [TK]D-Fender | perf3kt, Just look at the extremely limed call flow capabilities that FreePBX gives you. There you have it. Not even a multi-tennent capability. How about call-back systems? If I want to route calls based on time of day, weatcher in Istabul and divide by the score of the last Lakers game? |
| 04:54.12 | [TK]D-Fender | perf3kt, Yes, gui can be sold, but it undervalues YOU. |
| 04:54.45 | perf3kt | but there is a market for sellers and for supporters |
| 04:54.51 | perf3kt | and I thnk that is what the seperation is |
| 04:55.02 | [TK]D-Fender | perf3kt, well, we've layed out the cards, pick your hand an live with it. |
| 04:56.00 | [TK]D-Fender | perf3kt, buy if you have issues even satisfying *'s dependencies to install, then that might say something about what you should be considering. |
| 04:56.12 | perf3kt | lol |
| 04:58.31 | [TK]D-Fender | but* |
| 04:59.14 | [TK]D-Fender | perf3kt, Anyways enough diet-harsh reality for you.... |
| 05:00.13 | *** part/#asterisk tripppy (n=u@60-242-11-223.static.tpgi.com.au) |
| 05:00.30 | CrazyTux | Whats the special extension for invalid/timeout |
| 05:00.34 | CrazyTux | i, and t? |
| 05:00.40 | CrazyTux | version 1.4 |
| 05:04.46 | [TK]D-Fender | crazyTux : yes |
| 05:04.59 | [TK]D-Fender | crazyTux :hasn't changed. Ever |
| 05:05.18 | CunningPike | Someone really should document those |
| 05:05.24 | CunningPike | In a wiki or something |
| 05:05.28 | CunningPike | Or maybe a book |
| 05:05.32 | CrazyTux | CunningPike, http://www.voip-info.org/wiki/view/Asterisk+standard+extensions#Example |
| 05:05.34 | [TK]D-Fender | CunningPike, Or a book! I bet that'd sell! |
| 05:05.43 | CunningPike | [TK]D-Fender: ;-) |
| 05:06.39 | CrazyTux | You guys think I don't RTFM? I do, just when things don't work as expected, I just like to get a second opinion. |
| 05:06.59 | CunningPike | CrazyTux: Yes, but then ask question related to it! |
| 05:07.11 | CunningPike | Then folks can help |
| 05:07.55 | CrazyTux | Alright Question A), TIMEOUT(response) --- does this include AFTER audio for "Background", etc has been played, during, etc? |
| 05:08.14 | [TK]D-Fender | crazyTux : following the END |
| 05:08.28 | CrazyTux | Question B) -- why the hell is my TIMEOUT(response)=10 ... not hitting the t stdexten ? |
| 05:08.43 | CrazyTux | I get auto failthrough channel, status unknown.... |
| 05:08.59 | [TK]D-Fender | crazyTux : Guess you'd have to pastebin your whole IVR for us to validate. You don't think we TRUST you do you? :p |
| 05:09.15 | [TK]D-Fender | crazyTux : Well tahts because you left autofallthrough =yes! :p |
| 05:09.35 | [TK]D-Fender | crazyTuxand the second your exten ran out your call DIES |
| 05:09.48 | [TK]D-Fender | crazyTux : welcome to 1.2 :) |
| 05:09.52 | [TK]D-Fender | (let alone MORE) |
| 05:10.36 | [TK]D-Fender | NEXT!@!@ (c) BKW |
| 05:10.38 | JunK-Y | i dont really see the need of still using 1.2 |
| 05:10.44 | CrazyTux | Lets see here.... autofailthrough, what do you mean with the whole "trust" ? lol. I have no problem pasting this 10 line IVR, pretty simple. |
| 05:10.50 | [TK]D-Fender | JunK-Y, just stating when this change came in |
| 05:11.38 | [TK]D-Fender | crazyTux : no need. your fallthrough says that you did not set "autofallthrough=no" in [general] |
| 05:12.18 | [TK]D-Fender | crazyTux : So go change it and apply your changes. |
| 05:15.24 | CrazyTux | [TK]D-Fender, nothing changes |
| 05:15.32 | CrazyTux | [TK]D-Fender, same results |
| 05:16.38 | CunningPike | CrazyTux: pastebin your dialplan |
| 05:16.40 | CunningPike | ~pb |
| 05:16.41 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 05:17.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:17.13 | [TK]D-Fender | CunningPike, no need. I've already pointed out the culprit immediately. |
| 05:18.43 | CrazyTux | [TK]D-Fender, actually, that was it :) |
| 05:18.47 | CrazyTux | [TK]D-Fender, thanks. |
| 05:19.02 | CrazyTux | [TK]D-Fender, I thought originally you meant set it TO yes. |
| 05:19.42 | [TK]D-Fender | [TK]D-Fender> crazyTux : Well tahts because you left autofallthrough =yes! :p <--- nope |
| 05:19.49 | [TK]D-Fender | "left" |
| 05:19.59 | [TK]D-Fender | I hide things... in the big print ;) |
| 05:20.01 | CrazyTux | [TK]D-Fender, yea, :) -- realized that. |
| 05:20.02 | CrazyTux | lol |
| 05:20.34 | CunningPike | :) |
| 05:20.38 | CunningPike | All's well that ends well |
| 05:22.32 | *** join/#asterisk the007killer (n=the007ki@61.29.2.98) |
| 05:23.26 | [TK]D-Fender | alrighty... I'm off to get a decent nights sleep |
| 05:23.30 | [TK]D-Fender | bbl |
| 05:23.32 | CrazyTux | [TK]D-Fender, sleep tite. |
| 05:23.40 | CrazyTux | [TK]D-Fender, don't let the bed bugs bite. |
| 05:23.54 | *** part/#asterisk Odie_flocon (n=odie@S0106000f3d5eae81.cg.shawcable.net) |
| 05:29.55 | *** join/#asterisk Knobber (n=sdf@111.069.dsl.mel.iprimus.net.au) |
| 05:32.07 | Knobber | Hello, I have a problem, there is probably a very simple solution, but it is driving me crazy. I am writing an AGI script to rate calls when 'h' extension is called. The only thing I can't pass to the script is the number dialed prior to the h extension being called. Ideas? |
| 05:38.30 | Knobber | Anyone? |
| 05:45.48 | JunK-Y | are you running DeadAGI? |
| 05:46.12 | JunK-Y | u need to ignore sighup |
| 05:50.25 | Knobber | Yes I am running DeadAGI |
| 05:51.04 | Knobber | Everything works, but I am not sure what will pass the Dialled extension to the script. |
| 05:51.23 | Knobber | ${EXTEN} results in 'h' |
| 05:52.50 | Knobber | Say I dial '1234567', then hangup, I can pass dialed seconds, etc etc to my script fine, but not that actual dialed number, being 1234567. |
| 05:55.34 | JunK-Y | i dont understand what you are trying to do. |
| 05:56.34 | Knobber | I am trying to rate a call on hangup based on the number dialed. |
| 05:57.07 | Knobber | I cannot work out how to pass the number dialed to my AGI script once the 'h' extension has been called. |
| 05:57.58 | Knobber | I can pass the source, dialedtime, answeredtime etc, I just cannot pass the dst. |
| 05:58.08 | Knobber | As I am not sure how. |
| 06:02.18 | *** join/#asterisk zeeesh (n=zeeesh@202.166.161.45) |
| 06:10.47 | Knobber | I have just figured a work around, but I don't like it, as I believe there would be an easier way |
| 06:12.04 | Knobber | Work around is to create an astdb key of: ${EXTEN}/UNIQUEID when call is being passed |
| 06:12.46 | Knobber | then when h extension is called, to get the UniqueID from astdb, extract the destination number, then remove the key |
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| 06:26.55 | alephcom | an option is to not have it in the dialplan at all. |
| 06:27.10 | alephcom | We use an asterisk module to call a script when a call is hungup. |
| 06:27.27 | alephcom | It's entirely seperate from the dialplan but we can pass what we need for variables. |
| 06:32.58 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:38.52 | Knobber | alephcom: Do you have any examples of this script I could look at? |
| 06:39.21 | *** join/#asterisk rati (n=rati@124.125.255.24) |
| 06:39.21 | Knobber | alephcom: Module rather |
| 06:42.27 | alephcom | The source code for the module is gpled. I'm not sure who the original author is but there's an updated copy in the astpp sourcecode |
| 06:42.39 | alephcom | sf.net/projects/astpp |
| 06:42.49 | alephcom | the code is attributed there. |
| 06:43.06 | alephcom | I believe there's also a text doc outlining how to call it. |
| 06:43.25 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:45.58 | bintut | anyone tried a fax passthrough over iax2? |
| 06:46.28 | nestAr | i did it over SIP worked great |
| 06:46.39 | nestAr | never had a IAX device to try it with |
| 06:49.33 | bintut | nestAr: care to share how you did it? |
| 06:49.55 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:50.42 | *** join/#asterisk lesterc (n=lesterc@CPE-121-216-217-67.nsw.bigpond.net.au) |
| 06:52.34 | nestAr | bintut: bought a sipura ATA, set it up, plugged a fax machine into it, pointed a DID to that sip channel and started faxin' |
| 06:56.29 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 06:58.22 | bintut | nestAr: ah, i see.. thanks. =) |
| 07:01.02 | bintut | brb |
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| 07:07.34 | alephcom | Knobber: If you need a hand with it grab me sometime |
| 07:14.06 | zeeesh | i m unable to hear my voice msgs ... getting this msg "[Nov 27 11:57:33] WARNING[30321]: app.c:598 __ast_play_and_record: No audio available on SIP/702-09b67578?? |
| 07:14.07 | zeeesh | <PROTECTED> |
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| 08:05.26 | CrazyTux | Hey guys. |
| 08:05.37 | CrazyTux | Is there a way to do fileexist, type command, then action if true / false in asterisk? |
| 08:06.34 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 08:07.31 | cjk | hi, how can i configure my routes in a way that pbx/asterisk/pbxconfigurations/getfiles users controller pbxconfigurations and action pbx_asterisk_getfiles? |
| 08:07.37 | cjk | ups wrong channel |
| 08:07.38 | cjk | sorry |
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| 08:32.48 | zeeesh | voicemail.conf ... still unable to attach wave file .. getting msg "Recipient names must be specified"? what does it mean |
| 08:33.05 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:36.08 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 08:36.35 | zeeesh | "Recipient names must be specified" where shud i mention the recipient name ? |
| 08:39.33 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 08:39.37 | loompek | hi ppl |
| 08:39.42 | loompek | i've got a little ol question |
| 08:40.07 | loompek | SayNumber() only works for integers... not for floating point numbers |
| 08:40.15 | loompek | SayNumber("SIP/7-08217ae8", "274.75") |
| 08:46.50 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 08:57.47 | badcfe | is there a way of restraining allowed codecs for sip but without any preference order from asterisk? |
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| 09:03.37 | mort_gib | badcfe > disallow=all, allow=ulaw |
| 09:09.16 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
| 09:09.40 | Grnd-Wire | good evening! Does anyone know anything about Swift, or app_swift? |
| 09:11.09 | CrazyTux | He guys --- need some help with manager.conf / AMI / originating a call. |
| 09:12.26 | tzafrir | CrazyTux, we need some help helping you without specific details |
| 09:13.54 | *** join/#asterisk R1ck (i=rick@belphegor.deadlysins.nl) |
| 09:14.49 | Grnd-Wire | tzafrir: heh - If I give you specific details, can you help me with swift? |
| 09:16.02 | tzafrir | maybe |
| 09:16.16 | tzafrir | IIRC swift is already packaged with some distributions |
| 09:16.21 | tzafrir | e.g: with Debian |
| 09:16.33 | tzafrir | with others you may need to install it yourself |
| 09:16.38 | tzafrir | app_switf? |
| 09:16.47 | Grnd-Wire | tzafrir: I'm just getting started.. I am running CentOS 5.. I just isntalled the voice from Cepstral's website.. |
| 09:17.09 | Grnd-Wire | Well that's my question - some of this information seems REALLY old, so I don't know what to believe, and what will even still work with * 1.4 ? |
| 09:17.38 | CrazyTux | tzafrir, ok, I simply want to, originate a call to a local extension, and, connect the two. |
| 09:17.40 | Grnd-Wire | The app_swift instructions don't seem to make sense for 1.4 .. and it was written for Asterisk 1.0.x which makes me wonder. |
| 09:17.57 | CrazyTux | tzafrir, i.e. exten => foobar,1,Playback(some-audo) |
| 09:17.58 | tzafrir | hmm... I must have confused Swift with another porgram |
| 09:18.15 | Grnd-Wire | tzafrir: Text to speech stuff? |
| 09:18.27 | tzafrir | festival |
| 09:18.34 | tzafrir | there's also another one |
| 09:18.52 | Grnd-Wire | yeah, Cepstral's product is infinitely better than Festival sounds.. |
| 09:19.11 | Grnd-Wire | hmm - flite, but that is festival lite.. I would presume it sounds about as bad.. ? |
| 09:19.19 | R1ck | hello. I have a Siemens telephone system with about 20 stations, at what point in the telephone architecture should I place an Asterisk system if I want it to handle all my incoming/outgoing calls? |
| 09:20.06 | R1ck | can it be hooked up like any other station or should it come before the Siemens device |
| 09:20.35 | Grnd-Wire | R1ck: You would probably want to put it right in front of your Siemens.. interface to it with a T1 board.. |
| 09:20.57 | Grnd-Wire | The siemens will see it like the phone company.. |
| 09:20.59 | R1ck | Grnd-Wire: ehh, whats a T1 board? :) |
| 09:21.34 | Grnd-Wire | R1ck: wow.. You're going to need to get a consultant to work with you, especially since it will probably involve installing hardware in the siemens switch as well. |
| 09:22.05 | CrazyTux | tzafrir, i.e. I want to simply connect, an originated, end peer point, to an extension, all locally. |
| 09:22.17 | R1ck | Grnd-Wire: oh. I was hoping i could just install Asterisk on a server with a modem, and redirect all incoming calls to that modem's extension number |
| 09:22.29 | CrazyTux | tzafrir, so, I would want to connect user Foobar-UserA, to channel that has opened exten => foobar,1,PlayBack().... |
| 09:22.42 | mort_gib | R1ck modems wont work.... |
| 09:23.08 | Grnd-Wire | R1ck yah, you're talking about specialized voice hardware.. How many lines do you have coming into your current switch? |
| 09:23.28 | *** join/#asterisk Naeem (n=chatzill@62.240.47.161) |
| 09:26.24 | mort_gib | R1ck> You need special hardware to interface with your telco, if you want to handle all incoming outgoing calls, in which case the Asterisk server will have to sit in front of your Siemens PBX |
| 09:26.57 | R1ck | Grnd-Wire: about 5 i think |
| 09:27.20 | mort_gib | R1ck, you would then need special hardware to be able to pass calls from Asterisk to your Siemens box (Grnd-Wire is right) you might even need to change your Siemens setup |
| 09:27.29 | Grnd-Wire | R1ck: Why are you even wanting to do this? What do you hope to gain? What are your expectations? |
| 09:27.30 | R1ck | hmm |
| 09:27.55 | mort_gib | R1ck. I bet he wants to strat routing calls over VOIP :-) |
| 09:28.11 | CrazyTux | Anyone have any test examples of manager.conf / AMI / etc. |
| 09:28.19 | R1ck | I want users to be able to call from their pc's, and to accept calls on their pc's, and to have a menu kind of thing that customers calling us, have to go thru before they reach the right department |
| 09:28.21 | mort_gib | Most commercial PBX's charges you an arm and a leg for that! |
| 09:28.22 | Grnd-Wire | That is certainly a legitimate configuration.. |
| 09:28.52 | Grnd-Wire | R1ck; I think it's going to be important to contact someone who can work with you to design a system that will do what you need. |
| 09:29.34 | Grnd-Wire | R1ck: What you are asking for is certainly possible, reasonable, and sane.. but there's a lot of variables, and a lot of questions that need to handled by someone who is familiar with the specifics of your system, and the call flow in your office. |
| 09:29.35 | R1ck | Grnd-Wire: what would that person need to do? I want to learn how to work with Asterisk myself |
| 09:30.06 | R1ck | I can start with Asterisk handling just a single line |
| 09:30.28 | R1ck | could that be done with one modem, or do I still need "special" hardware? |
| 09:30.34 | Grnd-Wire | R1ck: Well, if you don't know what a T1 board is - I have doubts that you have a good handle on the technical portions.. PLUS - do you know how to change programming on your siemens? |
| 09:30.44 | R1ck | I do know a company I can call for the siemens related stuff |
| 09:30.56 | mort_gib | R1ck: there is a steep learning courve, selecting the right hardware is/can be uhm a challange |
| 09:31.04 | Grnd-Wire | R1ck: Absolutely not.. You're looking at about $200 for a single port card.. but it won't pass any caller ID information.. |
| 09:31.12 | R1ck | hmm |
| 09:31.27 | R1ck | Grnd-Wire: so what is this T1 card? I'm dutch, maybe its called something else here |
| 09:31.28 | Grnd-Wire | R1ck: You need to involve me from the START, because you'll be working around limitations in that switch.. |
| 09:31.43 | Grnd-Wire | R1ck: heh.. E1 card then.. |
| 09:31.54 | Grnd-Wire | ack.. Involve me.. I meant "em" |
| 09:31.59 | R1ck | well what does it do? |
| 09:32.31 | mort_gib | Carefull, you need to know how your local telco offers lines... |
| 09:32.51 | Grnd-Wire | R1ck: It lets you stick 29 (for an E1) conversations through a single little digital cable.. rather than needing a big bundle of cables (and ports) to do the same thing.. |
| 09:32.55 | R1ck | we just got this idea to want to do this, and I heard about Asterisk, so I'm just looking into the possibilities of doing this ourselves |
| 09:33.12 | mort_gib | E1's are fine but sometimes really expensive |
| 09:33.35 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 09:33.36 | *** join/#asterisk troy- (n=troy@DCC.SEND.startkeylogger.000.telephreak.org) |
| 09:33.57 | Grnd-Wire | R1ck: Like I said, it's a great idea.. but you absolutely must talk to someone who knows the entire equation.. If you get into thise without ALL of the facts, you will end up in a bottomless hole of a project - it'll keep taking money, time, and equipment.. and still not work the way management expects it to |
| 09:34.00 | mort_gib | R1ck> not a bad idea, but a consultant can fast track you to a working system, If you have the time you will be fine |
| 09:34.42 | Grnd-Wire | mort_gib: The siemens would interface to Asterisk through an E1 card I would hope.. How Asterisk gets to the PSTN is indeed a consideration for the consultant to address.. |
| 09:34.51 | R1ck | ok. well I gotta go in a meeting but i'll be back later :) |
| 09:34.58 | Grnd-Wire | ok.. |
| 09:34.58 | R1ck | thanks already for the time |
| 09:35.05 | R1ck | much appreciated |
| 09:35.10 | Grnd-Wire | ya |
| 09:38.25 | zeeesh | voicemail.conf ... still unable to attach wave file .. getting msg "Recipient names must be specified"? what does it mean |
| 09:38.36 | zeeesh | "Recipient names must be specified" where shud i mention the recipient name ? |
| 09:38.44 | mort_gib | Grnd-Wire: I'm not an expert on Siemens systems :-) |
| 09:39.00 | mort_gib | Sounds right though, also sounds expensive! |
| 09:40.08 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 09:40.33 | Dr-Linux | how can i send DTMF once call is bridged? |
| 09:41.31 | Grnd-Wire | mort_gib: Well you can get a dual T1 board for Asterisk for ~$600 or so.. The Siemens module could be two or three times that much after installation, and all of the configuration.. |
| 09:41.57 | Grnd-Wire | The best thing he can hope for is find someone who can program his Siemens that knows how to configure an Asterisk dialplan, and knows how to configure Zaptel devices.. :0 |
| 09:42.09 | Dr-Linux | question: what option i can use in Dial command to senf info. Any option other than CallerID and extension? |
| 09:42.16 | mort_gib | Agree! |
| 09:42.31 | Dr-Linux | Grnd-Wire: any advice? |
| 09:42.56 | mort_gib | But you are right! He needs help, only sometimes these project can only be sold if the local IT dept can do it all |
| 09:43.17 | mort_gib | -If we can control our PBX ourselves then we can... |
| 09:43.24 | mort_gib | Been there! |
| 09:43.40 | Grnd-Wire | mort_gib: heh.. yeah - Well that's the sort of services I offer to my clients.. I do voice, data, WAN, and conventional phone systems.. :) |
| 09:43.57 | Grnd-Wire | Dr-Linux: I'm not even seeing your question.. Hit me again.. |
| 09:44.19 | mort_gib | Grnd-Wire: Yeah? i do much the same, just started on Asterisk though.... |
| 09:44.44 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:45.10 | Dr-Linux | ok |
| 09:45.34 | *** join/#asterisk saftsack (n=saftsack@pD9E07A55.dip.t-dialin.net) |
| 09:47.40 | Dr-Linux | Grnd-Wire: i wanna send different infos with Dial command to other end, AFAIK, i can send CallerID and extension, what else option i can use to send info? |
| 09:48.35 | Grnd-Wire | Dr-Linux: Who are you calling? When calling someone on the PSTN, caller ID is the only thing you can change - and even then, most providers don't let you change that.. Some do though, I know Voipstreet will let you. |
| 09:48.42 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 09:49.06 | Dr-Linux | Grnd-Wire: i'm calling on SIP/host |
| 09:49.52 | Grnd-Wire | Dr-Linux: I'm still no tunderstanding you.. You're on a SIP device, and who are you calling? Is it another SIP device that is on your local Asterisk server? |
| 09:50.44 | Dr-Linux | Grnd-Wire: i've asterisk server and using sip i'm sending call to other SIP switch |
| 09:51.09 | Dr-Linux | hhm... |
| 09:51.15 | Dr-Linux | lemme give you example here: |
| 09:52.13 | Dr-Linux | exten => 333,1,Dial(SIP/host/${EXTEN}) |
| 09:52.43 | Dr-Linux | so before this i can set callerid and as callerID i can send some info/digits |
| 09:53.01 | Dr-Linux | then other thing is ${EXTEN} |
| 09:53.18 | Grnd-Wire | Are you asking if you CAN set other things, or are you TELLING me that you can? |
| 09:53.20 | Dr-Linux | that's what SIP switch will recieve |
| 09:53.39 | Dr-Linux | Grnd-Wire: i'm asking you |
| 09:53.53 | Grnd-Wire | Dr-Linux; ok.. |
| 09:54.08 | Dr-Linux | GreggB: bcoz i wanna send 3 type of infos to SIP switch |
| 09:54.14 | Grnd-Wire | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid |
| 09:54.23 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 09:54.31 | Dr-Linux | Grnd-Wire: i'm not that bad with such things, but mybe you could suggest me some good things |
| 09:54.42 | Grnd-Wire | Dr-Linux: Go there - at the bottom of the page they give an example of how to change your outgoing CID before you initiate the dial command. |
| 09:55.20 | Dr-Linux | Grnd-Wire: that's pretty easy to change - that's one option using what i can send info |
| 09:56.12 | Dr-Linux | i wanna send 3 parameters to SIP switch |
| 09:56.20 | Dr-Linux | callerid can do one for me |
| 09:56.50 | Dr-Linux | hhm.. |
| 09:57.14 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 09:57.17 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:57.35 | Dr-Linux | Grnd-Wire: can i use callerid(num) and callerid(name) both with different options |
| 09:57.50 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:58.08 | mort_gib | Grnd-Wire: Still here?? |
| 09:59.27 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:00.54 | Grnd-Wire | Dr-Linux: yes, that's the idea |
| 10:01.08 | Grnd-Wire | ok guys - Gotta go.. Have a good night! |
| 10:03.43 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:04.06 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 10:13.31 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 10:22.17 | yang | I experience the following problem when calling http://openpaste.org/en/4064/ |
| 10:25.14 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
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| 10:36.01 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 10:36.10 | yang | fixed |
| 10:46.51 | *** join/#asterisk E-bola2 (n=m@87.63.94.170) |
| 10:47.26 | *** join/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 10:47.40 | E-bola2 | Can anybody help me with the OSS/dsp channel? Im trying to use pickup2 to pick it up, but the channel state is 0 even if its ringing |
| 10:47.47 | E-bola2 | Doesnt channel states work for the console channel? |
| 10:48.09 | E-bola2 | Shouldnt it be in state 5 when its ringing? |
| 10:48.13 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 10:50.37 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
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| 10:56.41 | agx | Anyone had experiences with mobotix camera M10 and voip? it seems that does not respond to options so i've to turn off qualify; anybody can confirm? |
| 11:01.05 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:01.14 | CrazyTux | Hey guys |
| 11:01.36 | CrazyTux | Say I dial one of my users extensions, how can I switch, what response happens i.e. busy unavailable, etc. |
| 11:03.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:08.26 | mosty | use the ${DIALSTATUS} or (ugh) priority jumping |
| 11:15.15 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-259d97f1fef24013) |
| 11:18.39 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:25.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:32.11 | E-bola2 | can anybody tell me how i use the mixer comamnds in oss.conf to increase volume of the console channel? |
| 11:32.18 | E-bola2 | i cant find any examples or documentation on how to do that |
| 11:33.40 | loompek | umm |
| 11:36.16 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:37.32 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 11:37.34 | BockBilbo | hello |
| 11:39.50 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
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| 11:55.28 | BockBilbo | any of you know of any good voip provider wich outgoing callerid support? |
| 11:56.21 | ai-a | BockBilbo: eh, you might want to specify which country. |
| 11:56.31 | ai-a | then we can say we dont live in that country. |
| 11:56.44 | BockBilbo | xD |
| 11:56.44 | ai-a | and why not use google for researching this.. or your local telco / library for information |
| 11:56.51 | BockBilbo | Spain |
| 11:56.54 | BockBilbo | im using google |
| 11:56.57 | smithj | in soviet russia, contry specifies YOU |
| 11:57.07 | BockBilbo | and i've just found one provider |
| 11:57.08 | BockBilbo | :/ |
| 11:57.10 | smithj | (sorry, couldn't resist) |
| 11:57.14 | BockBilbo | hehe |
| 11:57.44 | smithj | o rly? |
| 11:57.48 | ai-a | Heh |
| 11:57.57 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 11:58.23 | smithj | not that i'm terribly concerned one way or the other, just a point of fact :) |
| 11:58.45 | ai-a | actaully it feels so cold here i could be in russia. |
| 11:58.55 | smithj | pft. you're talking to an alaskan |
| 11:59.00 | ai-a | im actually closer to BockBilbo. being in England |
| 11:59.35 | ai-a | BockBilbo: you can have free incomming landline numbers in uk / germany from www.sipgate.co.uk |
| 11:59.36 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 11:59.52 | smithj | BockBilbo: anyway. it being voip and all, another solution is to get a number in another country |
| 12:00.02 | BockBilbo | ai-a i dont need a landline number |
| 12:00.26 | ai-a | then use their voip outgoing pay/as/you/go |
| 12:00.50 | BockBilbo | right, but not all of the providers show the caller id |
| 12:00.51 | yang | I can offer free calls in the testing period, is anyone interested? |
| 12:01.15 | BockBilbo | i mean, when i call someone i want him to see on his phone that it's me who's calling |
| 12:02.48 | smithj | BockBilbo: thats why i was suggesting going with a non-spainish provider. your call will seem to be comming from, say, england or the united states but no one will care since to them it is an incomming call and "international |
| 12:03.00 | smithj | " voip calls are usually almost as cheap as "domestic" |
| 12:03.22 | smithj | er... spanish |
| 12:03.22 | zerocod3r | yang I need for testing purpose |
| 12:03.29 | BockBilbo | hehe |
| 12:04.22 | BockBilbo | smithj see... i've found a provider called Carpo, which doesn't have cheap rates but offers the opportunity to show your real telephone number as the call origin when making phone calls with them |
| 12:04.32 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 12:04.36 | BockBilbo | i need something similar but cheaper if possible |
| 12:04.43 | smithj | oh, you want to spoof it so that it shows your cell or landline or something? |
| 12:06.11 | BockBilbo | right |
| 12:06.35 | smithj | i don't know how much carpo is, but it may be cheaper to get whatever the lowest-cost voip is and then use a third party to spoof the caller id... someone like https://www.itellas.com/?searchengine=google&gclid=CNnph6L-_I8CFRlFQAodf1oXIQ |
| 12:06.39 | BockBilbo | i want it to show my landline number even though the phone is being made from other origin |
| 12:07.00 | smithj | $0.10/minute or $24/month. and the way the us dollar is now, thats about 0.10 euro |
| 12:07.09 | smithj | or maybe a wee bit more |
| 12:07.24 | BockBilbo | hehe |
| 12:08.26 | BockBilbo | mm that's not exactly what i want... |
| 12:08.30 | BockBilbo | but thanks anyway |
| 12:08.31 | BockBilbo | :) |
| 12:08.47 | smithj | well, i've go no other potentially useful advice |
| 12:08.50 | smithj | so i'll stfu :) |
| 12:12.45 | BockBilbo | thanks |
| 12:12.51 | hi365 | does anything need to be done to enable ajam at build time?? |
| 12:14.43 | hi365 | when i try to access my_ip/static-http/ajamdemo.html and click login i get error: 404: not found |
| 12:14.53 | hi365 | (asterisk 1.4.11) |
| 12:16.41 | *** join/#asterisk sebele67 (n=sebastie@194.169.203.240) |
| 12:17.22 | ai-a | hi365: read the user comments -. http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) |
| 12:17.38 | ai-a | also,,, googling ajamdemo 404 found loads of hits. |
| 12:27.30 | myiagy | hi, i'm having trouble here with an Audiocodes MP202, it replies 400 Bad Request when asterisk tries to send mwi notify to it.. i checked http://bugs.digium.com/view.php?id=8575 |
| 12:28.17 | myiagy | then i removed the (0/0) in chan_sip.c, recompiled, did a debug, the notify is exactly like the bug report says it should be to work with cisco.. apparently, that's not enough for this audiocodes |
| 12:28.54 | myiagy | mailbox= is set for all sip accounts |
| 12:29.18 | myiagy | any ideas? |
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| 12:33.45 | *** join/#asterisk coppice (n=chatzill@8.194.17.210.dyn.pacific.net.hk) |
| 12:36.50 | Dr-Linux | hi guys |
| 12:42.21 | *** join/#asterisk Champi (i=Champi@rootshell.fr) |
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| 12:52.36 | agx | is there a way to have a different SipAddHeader for every forked call when using Dial or Page with "SIP/snom&SIP/grandstream&SIP/sipura&SIP/yuxin&SIP/mobotix" ? i wish to set a specific Ring Tone for every different branded model on calls incoming from external |
| 13:02.38 | R1ck | so, i'm back ;) |
| 13:03.17 | tzafrir | coppice, even with a simple sample A4 tiff fax, I get a fax (through a zaptel loopback) hung forever |
| 13:03.33 | tzafrir | audio keeps getting sent |
| 13:03.49 | tzafrir | I got a 12MB sound file from monitor |
| 13:03.58 | tzafrir | mostly in one direction |
| 13:04.10 | tzafrir | before I used soft hangup |
| 13:08.21 | coppice | that's sad |
| 13:09.36 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 13:10.38 | mosty | agx, you could do that with chan_local |
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| 13:10.46 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-259d97f1fef24013) |
| 13:10.48 | *** join/#asterisk sandorp (n=sandor@dhcp-146.phx3.llnw.com) |
| 13:11.06 | agx | mosty yes, wanted to avoid have a context for "intercom calls" and "external ring calls" |
| 13:11.30 | zeeesh | configuring voicemail still could not get success to send wave file as an attachment to somewhere@hotmail or somewhere@yahoo.com ... right now getting this erro "[Nov 27 18:16:23] NOTICE[467]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/ivan-082805a8' not posted" and "[Nov 27 18:16:37] WARNING[467]: app.c:598 __ast_play_and_record: No audio available on SIP/test-b7d0a350??"? |
| 13:11.36 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:11.51 | mosty | agx: normally you set the ringtone on the phone itself |
| 13:12.27 | mosty | zeeesh, what codec is the call? |
| 13:13.25 | *** join/#asterisk macros73 (n=cs@dsl093-063-226.pit1.dsl.speakeasy.net) |
| 13:13.53 | zeeesh | <mosty>: if u asking in sip.conf .. i think i did't specified there .. shud i specified alaw or gsm ? |
| 13:14.11 | agx | mosty, i already have distinctive ringtones but the syntax is different for every brand of phone :) |
| 13:16.36 | mosty | agx: if you want uniform behaviour with something that's non-standardised then i recommend you pick one brand/model/range of phone and stick to that |
| 13:17.34 | agx | mosty, hehehe, impossible, you should know |
| 13:17.58 | mosty | then you will need to have a messy dialplan, most likely |
| 13:18.53 | agx | mosty, n.p. i don't have a dialplan, just 3 AGI that handle all |
| 13:19.20 | mosty | s/dialplan/agi/ then |
| 13:23.11 | loompek | is it possible for asterisk to connect with netcentrex? |
| 13:23.31 | mosty | what is netcentrex? |
| 13:23.50 | loompek | umm |
| 13:24.09 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:24.20 | loompek | an AT&T of some sort... |
| 13:24.20 | zeeesh | voicemail configuration: getting 1st error " WARNING[652]: chan_sip.c:2963 sip_call: No audio format found to offer. Cancelling call to ivan "? |
| 13:24.45 | loompek | i keep getting |
| 13:24.48 | loompek | [Nov 27 14:20:37] NOTICE[13543]: rtp.c:1279 ast_rtp_read: Unknown RTP codec 96 received from [...] |
| 13:26.59 | mosty | loompek, what codec is it configured to use? |
| 13:27.10 | *** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 13:28.57 | mosty | zeeesh, which codecs do you allow for that sip account? |
| 13:29.03 | *** join/#asterisk NirS (n=chatzill@84.94.21.64.cable.012.net.il) |
| 13:29.15 | zeeesh | <mosty>:ulaw gsm |
| 13:30.37 | zeeesh | <mosty>: sorry just "ulaw" and "alaw". |
| 13:30.57 | mosty | and what codecs does the sip client support? |
| 13:32.49 | zeeesh | <mosty>: using xlite and i think its supports both |
| 13:33.41 | NirS | hey all |
| 13:33.43 | NirS | I'm having a really funky AGI problem |
| 13:33.48 | NirS | anyone with good AGI experience ? |
| 13:35.18 | loompek | mosty alaw |
| 13:37.11 | [TK]D-Fender | zeeesh: by this point you shuold already have pastebined a failed call with SIP debug enabled so that we can see whats happening. |
| 13:38.05 | zeeesh | ok |
| 13:42.16 | zeeesh | voicemail configuration error: This content is stored as http://sial.org/pbot/28887 |
| 13:46.31 | [TK]D-Fender | zeeesh: try AGAIN. |
| 13:46.44 | *** join/#asterisk bantu (n=Miranda@p54A33653.dip0.t-ipconnect.de) |
| 13:48.29 | *** join/#asterisk awk (n=awk@kia.inet-corp.com) |
| 13:48.53 | awk | hmm, anyone had issues with sangoma card and pass through.. the ability to get a clean signal for fax to mail? |
| 13:49.01 | awk | I never get any delivery with pass through |
| 13:49.33 | cjk | hi, is it possible to make junghanns bri card run with the misdn channel driver? |
| 13:49.39 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 13:49.45 | awk | yes why would u want to though? |
| 13:49.49 | awk | rather use bristuff |
| 13:49.58 | VJFROMGT | can someone tell me what the "outisbusy" command does? |
| 13:50.00 | awk | or use a sangoma bri card then u dont need to use bristuff and can just use wanpipe |
| 13:50.04 | cjk | awk, bristuff just sucks. no honestly its a pain in the ass to get it compiled |
| 13:50.15 | awk | cjk i dont have that issue |
| 13:50.16 | awk | i find it easy |
| 13:50.19 | awk | and use it all the time |
| 13:50.29 | cjk | i used it on gentoo and worked great |
| 13:50.54 | badcfe | im doing more than one invokations of Dial() in my dialplan. is it possible to get a CDR created for each of tham? |
| 13:50.55 | cjk | but now on debian it causes problems |
| 13:51.14 | awk | on a polycom phone is there a way to see if somebody is logged in as an agent or not |
| 13:51.25 | awk | cjk wow on debian its a s simple as ever |
| 13:51.30 | badcfe | i now have a cdr-custom but there is only one CDR per incoming call (access), not per invokation of Dial |
| 13:51.33 | awk | I build those boxes on a daily basis |
| 13:51.38 | [TK]D-Fender | awk: A few hackish ways |
| 13:51.55 | awk | [TK]D-Fender oh? know of a site i can see these ways |
| 13:52.01 | awk | really need to try find out... |
| 13:52.35 | [TK]D-Fender | awk: 1. use the microbrowser. 2. Make a script that polls your agents and updates a Custom DeviceState flag watched by Presence. |
| 13:52.53 | JT | how rewarding it is to rebuild your laptop |
| 13:52.55 | JT | ...eventually |
| 13:53.12 | JT | that was at least 40 or 50 screws :/ |
| 13:53.48 | cjk | awk, ok how do you do on debian |
| 13:53.54 | cjk | which kernel, which bristuff? |
| 13:53.55 | loompek | does asterisk support h323? |
| 13:54.06 | awk | [TK]D-Fender have you got a scipt that does this and can pickup the hints? |
| 13:54.16 | awk | cjk every kernel ive used |
| 13:54.27 | [TK]D-Fender | awk: You use normal buddy watch to pick up hints. |
| 13:54.37 | awk | just make sure you have the header files installed and sym link it to /usr/src/linux-2.6 |
| 13:54.55 | awk | [TK]D-Fender so that will be able to pickup a hint if a agent is logged on or not |
| 13:55.04 | cjk | awk, you use 1.2 or 1.4? |
| 13:55.10 | awk | cjk both |
| 13:55.19 | awk | but rather stick to 1.2 for now as bristuff are slow on updates |
| 13:55.24 | awk | and have a stable 1.2.24 release finally |
| 13:55.36 | VJFROMGT | where can i edit my macros? |
| 13:55.41 | awk | cjk in future use a sangoma bri card and u wont have that issue with using bristuff again |
| 13:55.53 | cjk | awk, sangoma has bri cards? |
| 13:55.56 | awk | VJFROMGT well in your extensions file |
| 13:55.58 | awk | cjk yes... |
| 13:56.05 | awk | they new but work great |
| 13:56.33 | [TK]D-Fender | brb |
| 13:56.53 | cjk | awk, what about the digium bri? |
| 13:56.59 | awk | never knew they had one |
| 13:57.06 | badcfe | apparently the CDR gets posted when the calling channel hangsup. i want it generated once the Dial application finishes. how may i accomplish that? |
| 13:57.26 | NirS | hey all |
| 13:57.37 | awk | badcfe why? |
| 13:57.46 | NirS | anyone encountered an issue with AGI not being able to run AGI commands? |
| 13:57.47 | NirS | in version 1.4.14 ? |
| 13:58.03 | mosty | badcfe, when the calling channel hangs up, the dial command does finish, doesn't it? |
| 13:58.06 | badcfe | awk: because i have one call into * that does more Dial |
| 13:58.07 | awk | 1.4.14 is a fup of a release with a fup with rt not working properly |
| 13:58.11 | VJFROMGT | awk > I want to edit the macro itself, ie, i want the macro "outisbusy" act differently |
| 13:58.31 | awk | badcfe huh? |
| 13:58.33 | badcfe | awk: and i need facturation for each of his Dial |
| 13:58.39 | NirS | awk, do you suggest downgrading ? |
| 13:58.47 | awk | badcfe u mean a transfer call that initiates dial twice yes |
| 13:59.13 | badcfe | no transfer(), its dial() |
| 13:59.22 | awk | im talking about a transfer call |
| 13:59.25 | awk | it iniates dial twice |
| 13:59.33 | awk | if you want to view it realtime use asterisk manager |
| 13:59.37 | awk | and grab the cdr's from there.. |
| 13:59.40 | badcfe | he does dial() and when the callee hups, he may initiate yet a dial() to another target |
| 14:00.07 | mosty | badcfe, that is two calls, not one, isn't it? |
| 14:00.16 | awk | NirS: not sure.. I had to on a few sites.. due to a number of issues |
| 14:00.22 | badcfe | i need CDR generated for each one of his dial() |
| 14:00.23 | cjk | awk, do you know what this is ? /root/bristuff-0.4.0-test4/zaptel-1.4.4/wctdm24xxp.c: In function ‘wctdm_init_one’: |
| 14:00.33 | NirS | well, lets try it out |
| 14:00.58 | awk | badcfe use the manager then... |
| 14:01.00 | awk | telnet to it |
| 14:01.04 | badcfe | mosty: thats what i think too, but as its only one incoming call navigating in the dialplan, only _one_ CDR is generated |
| 14:01.29 | awk | cjk I need to a bit more of the error than that... |
| 14:01.33 | mosty | badcfe, when the caller hangs up, that is the end of the call right? |
| 14:01.39 | badcfe | yes. |
| 14:01.50 | badcfe | but i also want a CDR when dial() is done |
| 14:02.07 | badcfe | thats is .. when the callee hangs up |
| 14:02.18 | mosty | for what purpose? |
| 14:02.25 | awk | so ffs use the nocdr function then |
| 14:02.36 | awk | if you want to limit what u want |
| 14:02.49 | badcfe | cause my dialplan is so that in one incoming call the caller may invoke more dial() sequentialy |
| 14:03.07 | mosty | awk, badcfe wants two cdr's for a single call, not zero |
| 14:03.49 | cjk | awk, ok, im trying to compile zaptel can i query you? |
| 14:03.52 | awk | mosty so use 2 custom cdr's.. and pipe them.. |
| 14:03.52 | badcfe | yes. for one single incoming call yes. cause the dialplan is surch that more dial() may be effectuated and must be facturated each of them |
| 14:04.22 | badcfe | custon-cdr is up and running, but the CDR pops up only when the caller hups. not per dial() as i need |
| 14:04.35 | awk | badcfe use the manager |
| 14:04.41 | awk | cjk dont fucking flood me |
| 14:04.43 | awk | use pastebin |
| 14:04.45 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:05.44 | cjk | awk, http://asterisk.pastebin.ca/799258 |
| 14:12.33 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 14:18.22 | awk | what does ls -al /usr/src show? |
| 14:18.30 | awk | does it have a sym link to your kernel? |
| 14:21.29 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) [NETSPLIT VICTIM] |
| 14:22.14 | *** join/#asterisk stse (n=stse@p54A5B35D.dip0.t-ipconnect.de) |
| 14:27.08 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:27.10 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:27.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:27.28 | flujan | guys, someone running 1.4.14 on slackware 12? |
| 14:36.24 | awk | do people still use slackware? |
| 14:36.54 | coppice | people still use DOS |
| 14:37.59 | *** join/#asterisk techie (n=techie@adsl-76-214-16-246.dsl.lsan03.sbcglobal.net) |
| 14:38.52 | alrs | I went to a job interview last month where they were deploying rails apps on slackware |
| 14:39.02 | coppice | I have 8" floppies of that, if you need them |
| 14:39.34 | coppice | I remember seeing rails applications on DOS. British Rail's applications |
| 14:39.40 | alrs | it was only semi-weird, since the rubygems people are semi-hostile to anyone's package management other than their own |
| 14:39.41 | *** join/#asterisk techie (n=techie@adsl-76-214-16-246.dsl.lsan03.sbcglobal.net) |
| 14:44.29 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 14:44.58 | sandorp | can someone help me figure out why call transfers are not working for me when I use x-lite to answer a call? it looks like asterisk is not listening for the buttons pressed on x-lite; I thought I'd made the changes necessary to make it listen to DTMF, but am obviously wrong: http://www.pastebin.ca/799221 |
| 14:47.13 | *** join/#asterisk stse (n=stse@p54A5B35D.dip0.t-ipconnect.de) |
| 14:47.34 | *** join/#asterisk kkjoe (n=kkjoe@p57A687DA.dip0.t-ipconnect.de) |
| 14:50.12 | [TK]D-Fender | sandorp: go verfiy taht DTMF is picked up at all from your X-Lite. |
| 14:50.20 | [TK]D-Fender | kjashwdsasd typing failure |
| 14:51.23 | sandorp | [TK]D-Fender: how would I check? I know I can make outbound calls |
| 14:51.28 | *** join/#asterisk freezey (n=freezey@maher.mercy.edu) |
| 14:51.31 | codefreeze | zeeesh: you can ignore the not posted message. single-channel cdrs, unanswered, aren't posted. Just put in a fix involving an extra config file opt for this. |
| 14:52.02 | [TK]D-Fender | sandorp: Access voicemailmain or swomething. Dialing a call doesn't mean anything. your phone passes taht exten direct and has nothing to do with DTMF |
| 14:52.09 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:52.09 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:53.24 | [TK]D-Fender | sandorp: and please pastebin your features.conf |
| 14:54.20 | sandorp | [TK]D-Fender: I see entries when I dial login/password for voicemail |
| 14:54.40 | [TK]D-Fender | sandorp: So you can fully cruise the VMM menu? |
| 14:54.48 | stse | Hi! I'm using Asterisk 1.4 and have snom IP phones (320 and 360). Hints are registered in extensions.conf. If I subscribe to a certain SIP number with my Snom, I see the light blinking, if someone calls the number, or a steady light, if this person is talking. What I would like now is, that I can see the number of the caller in the display and can pickup the call by pressing the blinking light. Any hints how I can do this? |
| 14:54.53 | sandorp | yes |
| 14:55.04 | sandorp | I can pick up voicemail, delete it, etc. |
| 14:55.18 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:55.21 | [TK]D-Fender | sandorp: Ok, please pastebin your features.conf now. |
| 14:56.10 | sandorp | http://www.pastebin.ca/799299 |
| 14:56.20 | [TK]D-Fender | stse: For that you'd have to modify your extens so taht it checks if the device is ringing first and then do a "pickup" of some sort (look this up on the WIKI). |
| 14:56.54 | [TK]D-Fender | stse: 1 app that might help is "ChanIsAvail. I know you can use this to see if its in use, but I'm not sure about other states. |
| 14:57.02 | [TK]D-Fender | stse: Go take a look. |
| 14:57.11 | [TK]D-Fender | stse: Especially under the presence pages. |
| 14:57.43 | coppice | ah, the Mystic Meg pages |
| 14:59.38 | [TK]D-Fender | sandorp: do "core set debug 10". That should let you see DTMF being detected IIRC. |
| 15:00.01 | [TK]D-Fender | sandorp: Make sure * sees it coming in. Then you may have to check the speed at which you press the feature digits |
| 15:00.49 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:02.52 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 15:02.53 | stse | [TK]D-Fender: as I said, the light at the Snom *is* blinking. Manual pickup via the pickup extensions is working. |
| 15:03.22 | [TK]D-Fender | stse: the trick is to have * verify the ringing status when you dial their exten. |
| 15:03.48 | [TK]D-Fender | stse: Because I doubt you can tell your phone what do dial based on that state. |
| 15:04.01 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:04.26 | [TK]D-Fender | stse: this would be something actually do-able on an Aastra with a feature request I'd bet. |
| 15:04.50 | stse | [TK]D-Fender: I think there is something available in newer firmwares. For now I am more interested how I can display the caller id of the number in the display. |
| 15:05.13 | [TK]D-Fender | stse: CID of the person calling that exten? |
| 15:05.18 | [TK]D-Fender | (device rather) |
| 15:05.49 | sandorp | [TK]D-Fender: * is not seeing anything I press on x-lite; btw: the caller can hear the digits being dialed with x-lite |
| 15:06.21 | [TK]D-Fender | sandorp: Make sure * & x-lite are both on RFC2833 |
| 15:06.35 | *** join/#asterisk Strom_M (n=strom@m250e36d0.tmodns.net) |
| 15:06.58 | stse | [TK]D-Fender: yes. Let's say, Person A is calling Person B and I am subscribed to B's number. My snom light is now blinking, but I don't see who is calling Person B. |
| 15:07.23 | [TK]D-Fender | stse: oh boy.... no sane way on the phone I can think of.... |
| 15:08.40 | [TK]D-Fender | stse: I can picture all o fthis through a web-based console (FOP does all this already IIRC), but there is no phone-based way for that. |
| 15:10.57 | stse | [TK]D-Fender: There was something like this in * 1.2.x. When I had the old version running, in my scenario above my snom showed "From B to B" which was quite useless, but * was capable of doing this without big magic. |
| 15:11.13 | R1ck | if I want an incoming call to be handled by Asterisk, what kind of device do I need in my computer for an isdn line, will a isdn modem work or do you need a card like the Digium TE120P ? |
| 15:11.56 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:12.48 | dijungal | i have some polycom ip430s that keeps rebooting spontaneously, any reason why? |
| 15:13.12 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:15.38 | Mw3 | R1ck: TE120P is for PRI, do you need BRI or PRI ISDN ? |
| 15:16.22 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 15:16.28 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 15:16.34 | jstew | Greetings |
| 15:17.17 | R1ck | Mw3: ah. I guess I need BRI |
| 15:17.55 | jstew | I'm having major issues with my tdm400p and via chipset. I'm about to scrap this mobo and get a better one. Any recommendations (for socket AM2)? |
| 15:18.21 | Mw3 | R1ck: where are you? |
| 15:18.34 | R1ck | Mw3: Duiven, Netherlands |
| 15:18.37 | jstew | I'm kind of looking at mobos with the nforce 4 chipset. yea or nay? |
| 15:19.03 | Mw3 | R1ck: for BRI you can use digiums 4 port BRI card |
| 15:19.27 | Mw3 | R1ck: or you can get some cheap hfcpci or avm fritz 1 port card and try misdn |
| 15:20.56 | R1ck | Mw3: so, after getting it, Asterisk can use it to accept/redirect calls to different clients? |
| 15:22.12 | *** join/#asterisk techie (n=techie@adsl-76-214-16-246.dsl.lsan03.sbcglobal.net) |
| 15:22.50 | Mw3 | R1ck: yes. but if you have point-to-point BRI with more than 2 channels, then you will need a multiport card. you can find BRI cards on shop.beronet.com for example |
| 15:23.14 | *** part/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 15:23.32 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 15:23.39 | R1ck | Mw3: hmm, what exactly is meant by point-to-point? |
| 15:25.08 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 15:25.20 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
| 15:25.50 | sandorp | [TK]D-Fender: I tried adding a sip peer in sip.conf to force dtmf mode to rfc2833 per the docs I found; not sure what to change/check on x-lite side; still not working though |
| 15:26.41 | nny | If i wanted 101 to ring as well, how would I do so with "exten => s,2,Goto(people,100,1)".. sorry such an easy question I know.. |
| 15:26.57 | nny | feel stupid for having to ask |
| 15:27.36 | [TK]D-Fender | nny: "core show application dial" |
| 15:28.15 | nny | [TK]D-Fender: in console? |
| 15:28.37 | [TK]D-Fender | nny: yes |
| 15:28.38 | nny | [TK]D-Fender: nm ty |
| 15:28.40 | [TK]D-Fender | nny: * CLI |
| 15:32.22 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-185.static.twtelecom.net) |
| 15:32.54 | nny | [TK]D-Fender: i have the book with me as well. I thought there was something I could do to exten => s,2,Goto(people,100,1) |
| 15:32.54 | nny | to have it ring both "people" 100 and 101 at the same time |
| 15:33.04 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:33.07 | nny | but I can't find it atm |
| 15:33.13 | [TK]D-Fender | nny: that is not "simultaneous. That is AFTER |
| 15:33.16 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:33.35 | [TK]D-Fender | nny: go read DIAL's instructions for how to ring multiple DEVICES simultaneously. |
| 15:33.35 | nny | so how do I make it simultaneous? |
| 15:33.49 | nny | nm |
| 15:33.51 | *** part/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 15:33.54 | sandorp | nny: are you looking for Dial(people&100&101,,) |
| 15:35.50 | sandorp | [TK]D-Fender: I tried adding a sip peer in sip.conf to force dtmf mode to rfc2833 per the docs I found; not sure what to change/check on x-lite side; still not working though; was adding a peer to sip.conf the right thing to do on the * side? |
| 15:36.06 | [TK]D-Fender | sandorp: and on "debug 10"? |
| 15:36.25 | sandorp | yes, I have debug set to 10 |
| 15:36.57 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:37.05 | [TK]D-Fender | (core debug) |
| 15:38.52 | dijungal | hi, is it a problem to connect 20 of my phones in the office to asterisk on the same 5060 port? |
| 15:39.11 | dijungal | will this cause any audio issues or signaling issues? |
| 15:39.17 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
| 15:39.41 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 15:40.00 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 15:41.11 | __freedom__lover | hey, someone knows how can i detect a calling that the callee pays the called? |
| 15:41.14 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 15:42.32 | sandorp | [TK]D-Fender: * just isn't seeing anything I press with x-lite after x-lite answers; my sip.conf snippet and log: http://www.pastebin.ca/799336 |
| 15:44.38 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 15:46.02 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 15:46.41 | *** join/#asterisk funkyhippy (n=notahipp@87.127.49.162) |
| 15:48.55 | funkyhippy | Hi Guys, I've just installed Asterisk Business Edition. How do I go about updating the install? Can't see any info in the manual just mentions you get regular updates |
| 15:49.40 | funkyhippy | and there seems to be no package manager installed |
| 15:50.21 | sandorp | funkyhippy: not sure this is the right forum for that question since that "package" is a digium product, if I recall correctly |
| 15:50.24 | [TK]D-Fender | funkyhippy: Something you should ask Digium support as thats a ditro question and you paid for it |
| 15:51.39 | funkyhippy | ok its just that in the docs they list this channel as means of support! lol never mind I'll try digium cheers. |
| 15:52.09 | *** join/#asterisk Trionnis (n=blah@209.201.67.250) |
| 15:58.30 | *** join/#asterisk dlynes (n=dlynes@d154-20-45-103.bchsia.telus.net) |
| 16:06.03 | variable_office | i tried recording and combining calls into one file, but the two different channels would always be off kilter |
| 16:06.10 | variable_office | any ideas? |
| 16:08.11 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:08.19 | [TK]D-Fender | variable_office: How are you doing it? |
| 16:10.02 | variable_office | [TK]D-Fender, Monitor(wav,/home/apsadmin/${TIMESTAMP}:${CALLERID(num)}:${EXTEN}) |
| 16:10.22 | variable_office | then i had the one flag to merge, but i turned it off |
| 16:10.26 | [TK]D-Fender | variable_office: "show application mixmonitor" <- |
| 16:10.41 | sandorp | [TK]D-Fender: I tried using SJPhone softphone and it claims to be using RFC2388; caller can still hear every digit dialed on softphone; seems * is just not listening |
| 16:11.16 | *** join/#asterisk quelo (n=quelo@host127-119-dynamic.181-80-r.retail.telecomitalia.it) |
| 16:11.40 | [TK]D-Fender | sandorp: Ok, I'm out of ideas for the moment... |
| 16:11.44 | Trionnis | kinda like my wife |
| 16:11.46 | Trionnis | never listens |
| 16:11.58 | variable_office | [TK]D-Fender, that doesnt even show the combination flag |
| 16:12.01 | sandorp | [TK]D-Fender: thanks for trying |
| 16:12.10 | quelo | Hi |
| 16:12.16 | [TK]D-Fender | variable_office: this doesn't NEED one. |
| 16:12.23 | [TK]D-Fender | variable_office: MIXmonitor |
| 16:12.47 | variable_office | ah, is mixmonitor new to 1.4? |
| 16:13.05 | [TK]D-Fender | variable_office: No, 1.2 or perhaps earlier |
| 16:14.17 | *** join/#asterisk ManxPower (n=manxpowe@6.sub-75-200-2.myvzw.com) |
| 16:14.19 | _Sam-- | 1.0.9 and up |
| 16:14.31 | variable_office | [TK]D-Fender, does it work well? had any problems with it? |
| 16:14.44 | [TK]D-Fender | variable_office: works fine, no issues to date |
| 16:14.54 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:15.24 | variable_office | was the monitor combination flag known to have issues? |
| 16:17.04 | [TK]D-Fender | variable_office: dunno... it does call Sox, so your version may be a factor... |
| 16:18.04 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:19.03 | nny | [TK]D-Fender: after further research, I have my DIAL defined in a seperate area, which basically reads exten => s,1,Dial(${ARG2},20) where ARG2 is defined in [people] under exten => 100,1,Macro(stdexten,100,SIP/100) |
| 16:19.03 | nny | exten => 101,1,Macro(stdexten,101,SIP/101) |
| 16:20.21 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 16:20.59 | [TK]D-Fender | nny: So look at your dial in CLI as it gets called, and read dials instructions again to see how to ring multiple devices |
| 16:21.30 | *** part/#asterisk sandorp (n=sandor@dhcp-146.phx3.llnw.com) |
| 16:22.03 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 16:22.32 | nny | k reading |
| 16:23.33 | variable_office | [TK]D-Fender, how come it wasnt added with ${TIMESTAMP} |
| 16:23.57 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 16:24.18 | [TK]D-Fender | variable_office: because maybe that variable is deprecated..... |
| 16:24.27 | variable_office | appears to be |
| 16:25.10 | variable_office | any idea on the replacement? |
| 16:25.39 | nny | [TK]D-Fender: if i read this right, the ability is built in as defined by Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]): |
| 16:25.39 | nny | This application will place calls to one or more specified channels. |
| 16:26.00 | ManxPower | variable_office: There is a super secret document located in /path/to/src/asterisk-1.4/doc/channelvariables.txt BUT DON'T TELL ANYONE! |
| 16:26.15 | [TK]D-Fender | nny: Yes |
| 16:26.45 | variable_office | ManxPower, thanks for keeping me in the loop |
| 16:27.14 | nny | so Dial(${ARG2},20&(${ARG4},20))? |
| 16:27.21 | nny | meh thats not right |
| 16:27.23 | nestAr | lol |
| 16:27.43 | *** join/#asterisk yangvnc (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de) |
| 16:27.48 | nny | obviously regardless I need to define both prior to |
| 16:27.52 | ManxPower | variable_office: there are many secret documents there. The Asterisk Secret System Horde Order Linking Everyone has lots of docs. |
| 16:28.03 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-165-45.red.bezeqint.net) |
| 16:28.06 | nny | ManxPower: lol |
| 16:28.20 | ManxPower | nny: Try to get it working WITHOUT macros and variables forst. |
| 16:28.27 | variable_office | lol |
| 16:28.39 | Dovid | hi. out of no where my asterisk box started rejecting all inbound calls. i restart of asterisk fixed it but I am trying to understand why. The error i got in the CLI was: Insufficient information for SDP (m = '', c = '') |
| 16:28.55 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:29.00 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:29.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:29.03 | ManxPower | Dovid: what version of asterisk? |
| 16:29.18 | Dovid | 1.2.8 |
| 16:29.40 | Dovid | oops |
| 16:29.42 | Dovid | 1.2.18 |
| 16:30.46 | Dovid | ManxPower: It happend out of no where. I did not want to wait to see what the issue was so I restarted asterisk. I am trying to figure out what went wrong. |
| 16:31.34 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:31.40 | ManxPower | Dovid: upgrade to the latest 1.2.x before trying to diagnose the issue. |
| 16:32.43 | Dovid | ManxPower: I wish it was that simple. I dont have control over that (goto love corporate). |
| 16:32.52 | ManxPower | Dovid: it sucks to be you. |
| 16:32.54 | Dovid | i want to know why all of a sudden it happend. what caused it et. |
| 16:33.02 | ManxPower | I try not to manage a system I can't change. |
| 16:33.17 | ManxPower | because then it's not management, it's just suicide. |
| 16:33.37 | *** join/#asterisk nohup_ (n=nohup@crack.nohup.nl) |
| 16:33.40 | nohup_ | hello! :) |
| 16:34.21 | Dovid | ManxPower: I agree. Do you know of any issue in 1.2.18 ? |
| 16:34.31 | nohup_ | i'm looking for a way to call 2 lines, and then connect the two together... (it's supposed to become a site on which you can enter your own number, and a destination number... ) |
| 16:34.47 | nohup_ | and it'll be using SIP, connecting to my local asterisk server... |
| 16:34.54 | [TK]D-Fender | nohup_: lookup "call files" and "AMI originate" on the WIKI |
| 16:34.56 | ManxPower | Dovid: You would have to check the detailed changelog in 1.2.x latest to know if anything is obvious. |
| 16:34.57 | [TK]D-Fender | ~wikis |
| 16:34.57 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:35.31 | nohup_ | okay, i' |
| 16:35.34 | nohup_ | ll check that |
| 16:35.56 | Trionnis | hah |
| 16:36.00 | Trionnis | sounds familiar :) |
| 16:37.52 | [TK]D-Fender | Trionnis: shh ;) |
| 16:38.06 | Trionnis | :> |
| 16:38.18 | russellb | o.O |
| 16:39.15 | Trionnis | hi russell |
| 16:39.18 | Trionnis | how goes it |
| 16:40.16 | Trionnis | since you're here, are there any docs on the http interface to AMI? |
| 16:40.18 | *** join/#asterisk ZPertee (n=ZPertee@dhcp166-233.wireless.uakron.edu) |
| 16:40.27 | Trionnis | usage hints, etc? |
| 16:40.46 | russellb | heh, um ... |
| 16:40.47 | russellb | ~book |
| 16:40.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:40.57 | Trionnis | not in there, it's sitting on the desk behind me |
| 16:40.59 | Trionnis | :P |
| 16:41.02 | russellb | that's the best overall asterisk documentation that exists IMO |
| 16:41.26 | Trionnis | as usual, voip-info is wrong to |
| 16:41.32 | russellb | i can't in good conscience recommend anything other than that, and the stuff that is in asterisk itself |
| 16:41.47 | Trionnis | they're saying "show http" will give info, and it doesn't exist |
| 16:41.47 | russellb | as you said, the wiki is quite often misleading and just plain wrong |
| 16:41.50 | Trionnis | yup |
| 16:41.58 | ZPertee | I have been surfing the web and I stil haven't been able to figure out how to route incoming pstn calls through different contexts. I know how to do this with voip but not with pstn |
| 16:42.28 | russellb | ManxPower: lol .. |
| 16:42.29 | nestAr | the wiki is typically just right enough to point you in the direction you need to be.. |
| 16:42.33 | Dovid | russellb: What would cause asterisk to all of a sudden reject incoming calls and displays in the CLI: Insufficient information for SDP (m = '', c = '') |
| 16:42.36 | ManxPower | The Wiki is a useful resource as are the mailing list archives, but both of those tend to have much outdated and outright wrong information. You really need a good grounding in Asterisk before you use those resources. |
| 16:42.48 | Dovid | ManxPower: I am going through the change logs now |
| 16:42.55 | *** part/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 16:43.06 | jameswf | jbot: book |
| 16:43.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:43.43 | ManxPower | There have been days where I've sent 5 messages to the mailing lists, all correcting some major wrong information someone said. |
| 16:43.45 | jameswf | anyone have a free no windows needed ebook reader for blackberry? |
| 16:43.48 | nestAr | i think part of the problem of the wiki is just maintenance, due to constant changes in asterisk.. this is depreciated, this is is new, etc. etc.. |
| 16:44.09 | russellb | nestAr: yeah, it's tough to document a moving target |
| 16:44.43 | ManxPower | russellb: and yet the docs in asterisk-1.4/doc is amazingly up to date. |
| 16:44.56 | e` | how can I try to being troubleshooting poor voice quality? some users have been reporting crackling and poor voice quality on inbound/outbound calls |
| 16:45.12 | jameswf | ~qos |
| 16:45.13 | jbot | i heard qos is Quality of Service, a great source of information is located @ http://www.lartc.org |
| 16:45.32 | Dovid | russellb: Sorry to be a pest. did u see my question to u ? |
| 16:45.36 | jameswf | ~codec |
| 16:45.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:45.57 | Trionnis | wow, codec isn't in there? |
| 16:46.04 | *** join/#asterisk arpunk (n=0xc0ff33@200.118.170.145) |
| 16:46.08 | jameswf | yeah that sucks lol |
| 16:46.09 | arpunk | hi all |
| 16:46.11 | arpunk | how can I match a number that the local users can call (*9000 and also external users as 8099239000) ? |
| 16:46.46 | [TK]D-Fender | Trionnis: I'll get around to it :) I've done almost all of the others! |
| 16:47.01 | Trionnis | that's why I'm suprised |
| 16:47.02 | russellb | Dovid: don't know |
| 16:47.09 | Trionnis | I figured that would be one of the first ones you did |
| 16:47.11 | Trionnis | :) |
| 16:47.30 | [TK]D-Fender | Trionnis: No, its extremely rare anyone asks about that. |
| 16:47.45 | russellb | jbot: codec is muahahaha ... no useful information here! |
| 16:47.46 | jbot | ...but codec is already something else... |
| 16:47.49 | Dovid | russellb: I am trying to track it down. I dont fully understand that error. Any way you can explain it to me ? |
| 16:48.01 | Trionnis | um |
| 16:48.05 | russellb | Dovid: no, especially since you're not using a supported version |
| 16:48.16 | Trionnis | it's broken now |
| 16:48.23 | Trionnis | good job Russell |
| 16:48.28 | Trionnis | you killed it!! |
| 16:49.01 | [TK]D-Fender | russellb: I'll get around to that one shortly :) |
| 16:49.20 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 16:49.29 | Dovid | russellb: supported as in 1.4.X and up ? |
| 16:49.35 | jameswf | jbots dead? |
| 16:49.46 | russellb | Dovid: correct |
| 16:49.48 | jameswf | ~~ |
| 16:49.51 | jbot | Every moment in which I'm called upon is torture. |
| 16:49.58 | jameswf | ~codec |
| 16:50.10 | jameswf | ~audio |
| 16:50.11 | jbot | it has been said that audio is usually a codec issue. start with trying to set 'disallow=all' and 'allow=alaw' in sip.conf or the channel's config file if not using sip |
| 16:50.11 | russellb | ~thwack [TK]D-Fender |
| 16:50.13 | jbot | ACTION beats [TK]D-Fender on the eye with a UNIX Manual |
| 16:50.21 | [TK]D-Fender | :| |
| 16:50.24 | Trionnis | pwnt |
| 16:50.29 | Trionnis | by a machine, no less |
| 16:50.29 | jameswf | jbot: kill |
| 16:50.35 | russellb | ~hug [TK]D-Fender |
| 16:50.36 | jbot | ACTION jumps into [TK]D-Fender's lap and huggles and *hugs* [TK]D-Fender |
| 16:50.42 | Trionnis | erm... |
| 16:50.47 | Dovid | hehe |
| 16:50.55 | [TK]D-Fender | jbot: ~areyouadog ? |
| 16:50.56 | russellb | :-p |
| 16:51.05 | [TK]D-Fender | jbot: areyouadog |
| 16:51.06 | jbot | Bark! Bark! |
| 16:51.09 | [TK]D-Fender | :D |
| 16:51.18 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:51.29 | jameswf | jbot: kill newb |
| 16:51.30 | jbot | ACTION shoots a charged quark gun at newb |
| 16:51.48 | jameswf | jbot: eat poo |
| 16:51.49 | jbot | ACTION slurps up all the poo available |
| 16:51.53 | jameswf | lmao |
| 16:52.22 | Trionnis | ... |
| 16:52.26 | Trionnis | ...... |
| 16:52.42 | nny | so exten => s,1,Dial(SIP/100,20&SIP/101,20) |
| 16:52.42 | nny | Should call 100 and 101 right? |
| 16:52.56 | [TK]D-Fender | nny: Nope, read the formating again... |
| 16:52.58 | Qwell | no |
| 16:53.06 | Qwell | it'll call SIP/100 for 20&SIP/101 seconds |
| 16:53.07 | nny | k |
| 16:53.13 | Trionnis | haha |
| 16:53.14 | nny | haha |
| 16:53.14 | nny | oh |
| 16:53.17 | nny | man |
| 16:53.20 | nny | <--- shoot me |
| 16:53.26 | Trionnis | jbot: kill nny |
| 16:53.26 | jbot | ACTION shoots a ionized pseudomeson gun at nny |
| 16:53.26 | Strom_M | bang |
| 16:53.26 | Qwell | jbot: shoot nny |
| 16:53.27 | jbot | ACTION shoots nny in the foot with a phase pistol! |
| 16:53.31 | [TK]D-Fender | nny: Sorry... don't do guns any more :) |
| 16:53.34 | Trionnis | I was faster ;) |
| 16:53.38 | nny | lol yeah taze me |
| 16:53.40 | Qwell | [TK]D-Fender: jbot does |
| 16:53.54 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 16:54.59 | Trionnis | brb |
| 16:55.50 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:56.10 | Dovid | jbot: kill jbot |
| 16:56.10 | jbot | ACTION shoots a super-inverse fluxpositrino gun at jbot |
| 16:56.16 | Dovid | haha |
| 16:56.33 | Dovid | looks like suicide |
| 16:58.20 | nny | exten => s,1,Dial(${ARG2}[&SIP/101],20) |
| 16:58.26 | nny | that look any better? |
| 16:58.26 | nny | er |
| 16:58.35 | nny | exten => s,1,Dial(${SIP/100[&SIP/101],20) |
| 16:58.58 | nny | naop damn |
| 16:59.09 | nny | exten => s,1,Dial(SIP/100[&SIP/101],20) |
| 16:59.51 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 17:00.17 | nny | nope |
| 17:00.18 | [TK]D-Fender | nny: no []. Those shoulw you the order of optional parameters... |
| 17:00.20 | nny | spaces damnit |
| 17:00.24 | [TK]D-Fender | show* |
| 17:00.32 | nny | heh |
| 17:00.39 | nny | jbot: kill nny |
| 17:00.39 | jbot | ACTION shoots a inverse positrino gun at nny |
| 17:00.48 | [TK]D-Fender | nny: Ok, lets speed this u.. you're trying at least. exten => s,1,Dial(SIP/100&SIP/101,20) |
| 17:00.51 | [TK]D-Fender | up* |
| 17:01.06 | nny | heheh ty, i was so close |
| 17:01.07 | nny | thanks |
| 17:02.06 | nohup_ | i'm confused... do i need to tell asterisk to accept those files in /var/spool/asterisk/outgoing |
| 17:02.09 | nohup_ | ? |
| 17:03.53 | nohup_ | oh wait, it was a permissions thing :) |
| 17:05.38 | ZPertee | any suggestions for free/cheap sip DID |
| 17:08.16 | nohup_ | thanks, [TK]D-Fender.. wiki has been really helpfull :) |
| 17:08.26 | nohup_ | and now it's dinnertime! :) |
| 17:08.45 | *** part/#asterisk arpunk (n=0xc0ff33@200.118.170.145) |
| 17:10.43 | [TK]D-Fender | ZPertee: ... |
| 17:10.45 | [TK]D-Fender | ~cheap |
| 17:10.46 | jbot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 17:11.01 | [TK]D-Fender | ~ygwypf |
| 17:11.01 | jbot | hmm... ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 17:11.13 | Dr-Linux | i've two question about caller ID. 1, how many digits i can set as callerID, is there any limit? |
| 17:11.33 | Dr-Linux | 2, can i use callerid(num) and callerid(name) both with different options |
| 17:11.36 | [TK]D-Fender | Dr-Linux: on PST there is a limit. 12 or 16 I think |
| 17:11.42 | mort_gib | How good is FreePBX |
| 17:11.54 | jameswf | mort_gib: depends |
| 17:11.56 | [TK]D-Fender | mort_gib: good at WHAT is the question... |
| 17:12.07 | Dr-Linux | i see |
| 17:12.10 | Corydon76-dig | 15 chars on PSTN |
| 17:12.21 | Dr-Linux | [TK]D-Fender: can you help with my 2nd question? |
| 17:12.38 | yangvnc | These USA-TOOL FREE numbers are they reachable from all usa phones for free ? But is it also possible to call them from outside of USA ' |
| 17:12.41 | [TK]D-Fender | Dr-Linux: your 2nd question made no sense |
| 17:13.00 | [TK]D-Fender | yangvnc: You can never escape those tools... |
| 17:13.03 | mort_gib | Does it provide any functions that are really hard to get to work with Asterisk?? |
| 17:13.19 | [TK]D-Fender | yangdepends on your telco |
| 17:13.25 | mort_gib | In an easy way?? |
| 17:13.31 | Dr-Linux | [TK]D-Fender: lemme explain |
| 17:13.36 | [TK]D-Fender | yangvnc: And yes free in USA & usually North America |
| 17:13.46 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 17:13.50 | [TK]D-Fender | mort_gib: Some bits sure. |
| 17:13.51 | jameswf | mort_gib: hard is relitive, freepbx has limits but can be easier than doing all it does do by hand |
| 17:13.58 | nny | [TK]D-Fender: thanks for the help. I learn by trial by fire :) bbl, have a good day |
| 17:14.01 | Navion | ~security code |
| 17:14.03 | yangvnc | [TK]D-Fender: so its better to have one of those, the caller doesnt have to pay when calling |
| 17:14.06 | *** part/#asterisk stse (n=stse@p54A5B35D.dip0.t-ipconnect.de) |
| 17:14.15 | macTijn | oh! :( |
| 17:14.27 | macTijn | hm wait |
| 17:14.45 | Dr-Linux | [TK]D-Fender: i wanna send you two things with dial command on your SIP host i.e. billing number and callerid |
| 17:14.51 | mort_gib | Some of the functions that allows users to see what other users are doing seems to be a bit difficult (on call/DND) |
| 17:14.55 | [TK]D-Fender | yangvnc: Depending on a certain point of view |
| 17:15.10 | jameswf | if Your not an asterisk guy freepbx is good BUT is NO substitution for Learning, its like math you can use a calculator but you should know what to do without one |
| 17:15.27 | [TK]D-Fender | jameswf: not even so generous. |
| 17:15.41 | _x86_ | <PROTECTED> |
| 17:15.47 | _x86_ | what does the (1:0/1/0) mean? |
| 17:16.00 | [TK]D-Fender | mort_gib: FreePBX implement * along a very simple logic. If you want more than it does in terms of flexibility you could be screwed. |
| 17:16.10 | Dr-Linux | [TK]D-Fender: so i set callerid in first priority like 1,Set callerid (billing-number) and then in 2nd priority they callerid (name) |
| 17:16.11 | yangvnc | [TK]D-Fender: do you know which interface (manual) I could follow to be able to do something like when i call my asterisk extension (number) that it lets me enter an international number and establish a call to it - interlink ? |
| 17:16.30 | [TK]D-Fender | Dr-Linux: You can obviously set the name & number seperately. |
| 17:17.03 | [TK]D-Fender | yangvnc: go read... THE BOOK |
| 17:17.05 | [TK]D-Fender | ~book |
| 17:17.05 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:17.27 | [TK]D-Fender | yangvnc: You want to know how * works and what dialplan apps you have at your disposal, there's the book |
| 17:17.28 | Dr-Linux | [TK]D-Fender: great, and callerid(ani) as well? |
| 17:17.28 | yangvnc | yeah i got it, its only 650 pages |
| 17:17.31 | _Sam-- | sup Dr-L |
| 17:17.41 | [TK]D-Fender | Dr-Linux: All the same idea. |
| 17:18.01 | _x86_ | [TK]D-Fender: any ideas? |
| 17:18.33 | Dr-Linux | actually i wanna send 3 paramters with dial command, so this was an idea came in my mind, i just wanted to confirm |
| 17:18.39 | [TK]D-Fender | _x86_: I ignore little messages like that. I like seeing the line that CAUSED it, and debug to match. |
| 17:18.45 | mort_gib | I had a look at Trixbox, and didn't much like it... |
| 17:19.05 | jameswf | see mort_gib your allready fitting in |
| 17:19.09 | [TK]D-Fender | mort_gib: I think you missed the poitn... Trixbox = FreePBX + even more extra stuff |
| 17:19.22 | Dr-Linux | [TK]D-Fender: and i hope it will be easy to to parse at recieving end? if i'm not wrong |
| 17:19.26 | [TK]D-Fender | mort_gib: FreePBX is what is "bad " here. |
| 17:19.39 | [TK]D-Fender | Dr-Linux: parse? Parse with WHAT? |
| 17:19.54 | mort_gib | Yes, but keeping an open mind was what got me here in the first place... |
| 17:20.30 | Dr-Linux | [TK]D-Fender: like if i send you all these 3 info with different callerid options, will you be able to recive/handle all at your sip server? |
| 17:20.52 | jameswf | fonality has changed their slogan : "The art of exploitation, all the M$ stuff at half the price + monthly fees" |
| 17:21.08 | russellb | jameswf: lol! |
| 17:21.11 | mort_gib | Some features I haven't looked at yet, but I got most of the stuff that I normally hear people asking for working pretty fast... |
| 17:21.19 | [TK]D-Fender | Dr-Linux: give COMPLETE samples of whats on EACH END of the call. You keep talking about HALF of the story and asking me if everything will work the way you want. |
| 17:21.22 | Qwell | jameswf: may I quote you? |
| 17:21.34 | jameswf | for royalties :) |
| 17:21.42 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 17:21.42 | Qwell | How about free software? |
| 17:21.48 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 17:22.00 | russellb | Qwell: you should quit smoking |
| 17:22.06 | Qwell | I totally should |
| 17:22.08 | russellb | Qwell: it will help the cough |
| 17:22.13 | bkruse | is there a dialplan function to do space trimming? |
| 17:22.14 | Qwell | actually, it'll make it worse |
| 17:22.18 | Qwell | for a short time, anyhow |
| 17:22.18 | jameswf | I think we should all get pattents on other folks stuff.. |
| 17:22.26 | russellb | bkruse: maybe CUT() |
| 17:22.29 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 17:22.40 | Qwell | russellb: I got a free trial of this gum stuff... it kinda sucks :p |
| 17:22.41 | bkruse | russellb: hmm, i will check, ty |
| 17:22.44 | Qwell | the instructions were faulty |
| 17:22.51 | file | the user was faulty |
| 17:22.59 | Qwell | yeah, maybe.. so, get this |
| 17:23.02 | russellb | bkruse: it might not do it ... would be a good one to write though ;) |
| 17:23.09 | russellb | bkruse: ast_strip() ftw |
| 17:23.22 | Qwell | the instructions say to chew it, until you "feel a slight tingle", then you're supposed to put it between your gums/cheek |
| 17:23.25 | bkruse | russellb: will do :] |
| 17:23.29 | Qwell | problem is...it's cinnamon |
| 17:23.48 | Qwell | so it always tingles :p |
| 17:23.49 | apocn | Hello, I have an extension that users can dial in two ways (internal users dial to it using *9000 and external users 8299239000), how can I make it to match both without making 2 different extensions? |
| 17:24.00 | jameswf | if you have no teeth how do you quit smoking :) |
| 17:24.26 | Dr-Linux | [TK]D-Fender: i roughtly wrote 4 lines in notepad, i gonna paste 2 and then 2 lines here thats what i want |
| 17:24.28 | Qwell | I know a guy who quit smoking...triggered some genetic disease he had. |
| 17:24.39 | Qwell | there are only about 8 known cases of it O.o |
| 17:25.28 | Dr-Linux | [TK]D-Fender: let's say i wanna dial long distance calls through your server, and you want me to send 2 infos along the call |
| 17:25.31 | Dr-Linux | i'm sending you here |
| 17:25.35 | [TK]D-Fender | apocn: You can't. Those are 2 distinct numbers. |
| 17:25.37 | jameswf | sadly most people get lung cancer after they quit, so if you have smoked $25 years dont quit because prolonged smoking prevents cancer |
| 17:25.39 | Dr-Linux | exten => xx.,1,set callerid (num) |
| 17:25.39 | Dr-Linux | exten => xx.,2,set callerid (name) |
| 17:25.46 | Dr-Linux | two more... |
| 17:25.54 | Dr-Linux | exten => xx.,3,set callerid (ani) |
| 17:25.54 | Dr-Linux | exten => xx.,4,dial(SIP/Fender.com/${EXTEN}) |
| 17:25.59 | [TK]D-Fender | Dr-Linux: Go for it, the CODE isn't the issue |
| 17:26.14 | Dr-Linux | [TK]D-Fender: ignore errors |
| 17:26.21 | [TK]D-Fender | Dr-Linux: I asked what was on the OTHER END <------ |
| 17:26.35 | Dr-Linux | [TK]D-Fender: SIP |
| 17:26.49 | [TK]D-Fender | Dr-Linux: SIP is a PROTOCAL, not SOFTWARE! |
| 17:26.51 | [TK]D-Fender | sfdasfdlsdhgflkhsgfd |
| 17:26.56 | Dr-Linux | this is sip to sip communication, |
| 17:26.58 | Dr-Linux | hhm.. |
| 17:27.00 | Dr-Linux | good question |
| 17:27.05 | [TK]D-Fender | OMG... |
| 17:27.11 | [TK]D-Fender | must...not...kill...... |
| 17:27.14 | Dr-Linux | [TK]D-Fender: other end has SIP switch |
| 17:27.24 | [TK]D-Fender | Dr-Linux: how terminally vague. |
| 17:27.31 | Dr-Linux | [TK]D-Fender: i think other end dont have * |
| 17:27.33 | [TK]D-Fender | terminal..... |
| 17:28.50 | Dr-Linux | [TK]D-Fender: all i know about other end is "SIP Switch" |
| 17:29.12 | [TK]D-Fender | Dr-Linux: then "good luck". Looks fine from *'s point of view. |
| 17:29.48 | Dr-Linux | [TK]D-Fender: but this is good info i got from you that asterisk can send different callerID options in same call |
| 17:29.54 | Dr-Linux | hhm... |
| 17:30.09 | [TK]D-Fender | Dr-Linux: those are all parts of every call, of COURSE you can set them all. |
| 17:30.27 | Dr-Linux | but i wanted to make sure digit limit in callerID, if it's 12 or 16 :S |
| 17:30.39 | Dr-Linux | bcoz card have 16 digits |
| 17:30.54 | [TK]D-Fender | Dr-Linux: well I said on the **PSTN** You aren't even paying full attention to the answer. |
| 17:31.58 | Dr-Linux | [TK]D-Fender: i'm sorry i didn't understand PST |
| 17:31.59 | Dr-Linux | <[TK]D-Fender> Dr-Linux: on PST there is a limit. 12 or 16 I think |
| 17:32.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:33.07 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.6) |
| 17:33.15 | coolfreecode | hello :D |
| 17:33.27 | [TK]D-Fender | Dr-Linux: PSTN <- |
| 17:33.56 | Dr-Linux | so no limit on SIP? |
| 17:34.38 | [TK]D-Fender | Dr-Linux: less of a limit anyways. I'm sure each system has their own limit, but I don't know of a standard. |
| 17:35.00 | Dr-Linux | ok great, thanks ! |
| 17:35.34 | Dr-Linux | Qwell: cisco 7935 with asterisk? :P |
| 17:36.24 | [TK]D-Fender | Dr-Linux: First you're talking about *, then a "sip swith", and now a specific phone? Maybe it'll accept ALL the digits but only show you a limited amount. Who knows. How about you actuall go TRY something... |
| 17:37.26 | Dr-Linux | [TK]D-Fender: :) phone one is different question to Qwell and thanking you :) |
| 17:37.45 | Dr-Linux | [TK]D-Fender: since laster year i always buzz him with cisco 7935 phone |
| 17:39.07 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:39.41 | coolfreecode | hey guy That means: |
| 17:39.44 | coolfreecode | [Nov 27 06:36:29] NOTICE[5066]: chan_iax2.c:5258 register_verify: No registration for peer 'esclavo' (from 190.41.82.2) |
| 17:39.44 | coolfreecode | [Nov 27 06:37:02] NOTICE[5062]: chan_iax2.c:7951 socket_process: Registration of 'maestro' rejected: 'Registration Refused' from: '190.41.82.2' |
| 17:40.06 | apocn | Hello, I have an extension that users can dial in two ways (internal users dial to it using *9000 and external users 8299239000), how can I make it to match both without making 2 different extensions? |
| 17:40.34 | [TK]D-Fender | apocn: You can't. Those are 2 distinct numbers. |
| 17:40.35 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 17:40.39 | [TK]D-Fender | apocn: Already answered. |
| 17:41.07 | apocn | sorry, didnt see it |
| 17:41.48 | apocn | another question, when I use Agent/@1 (and of course I have defined group=1) it says invalid and doesnt work. |
| 17:43.22 | ManxPower | @1 is not a group |
| 17:44.31 | apocn | why not if I have defined it on the agents.conf file? |
| 17:45.27 | apocn | and on queues.conf I've done member => Agent/@1 |
| 17:46.44 | *** part/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 17:46.49 | Navion | Anyone know how to unforward an extension that was forwarded with *72 and won't unforward with *73? Asterisk CLI "database show" shows it CF'd. |
| 17:48.49 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 17:51.50 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 17:52.20 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 17:56.02 | ManxPower | Navion: that is more of a GUI question |
| 17:56.39 | ManxPower | For SIP phones, the phone can do the forwarding, for Zaptel Zap can do the forwarding, for everything else you have to write it in your dialplan |
| 17:57.24 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:57.59 | coolfreecode | hey guys is possible do calls client(SIP)--Asterisk--Trunk_IAX--Asterisk--client(SIP) |
| 17:58.09 | Strom_M | yes |
| 17:58.55 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:01.05 | Navion | ManxPower: The database is aterisk. I just read it with the CLI. What I need to know it how to fix the database if the station can't *73 and make it unforward. |
| 18:01.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:05.44 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:07.01 | [TK]D-Fender | Navion: "help database" from * CLI |
| 18:11.43 | errr | I have a problem with the builtin call monitor stuff from res_features.c in asterisk 1.2.24 When I get a call xfered to me and then I try to record the call the filename represents that who ever xfered me the call is who recorded it instead of me, is there any way to change this? |
| 18:11.47 | *** join/#asterisk masus (n=ethemc@88.248.73.2) |
| 18:12.10 | masus | hiaa , does anyone know how to change the port on audio codes ? |
| 18:12.26 | masus | i cant find the configuration from where to change it |
| 18:14.42 | *** join/#asterisk [N00B] (n=ckwall@206.71.78.172) |
| 18:14.55 | [N00B] | suddenly getting a lot of errors. can anyone possibly tell me what is going on? |
| 18:15.07 | [N00B] | pasting errors right now |
| 18:15.18 | [N00B] | http://pastebin.ca/799506 |
| 18:18.53 | *** part/#asterisk masus (n=ethemc@88.248.73.2) |
| 18:19.57 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:20.45 | [TK]D-Fender | [N00B]: 1st guess, you're using a wav in an unsupported format |
| 18:21.53 | nohup_ | what's that little tool which with you can 'listen' to your analog phone line on a ZAP interface again ? |
| 18:22.47 | [TK]D-Fender | nohup_: ZapBarge or ZhanSpy |
| 18:22.52 | [TK]D-Fender | ChanSpy* |
| 18:23.00 | Qwell | ZhanSpy...sounds...political |
| 18:23.20 | nohup_ | hmmm... |
| 18:23.34 | nohup_ | okay :) |
| 18:23.42 | nohup_ | i know i have one installed, but it's neither of those 2... |
| 18:24.02 | [TK]D-Fender | nohup_: feel free to tell us what it is when you find it. |
| 18:24.15 | [TK]D-Fender | nohup_: Oh, and there is ExtenSpy as well I gues |
| 18:24.17 | nohup_ | i'm searching :) |
| 18:25.36 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 18:25.46 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:26.24 | Navion | [TK]D-Fender: Can I use database del CF/<ext no> |
| 18:27.16 | [TK]D-Fender | Navion: No, I strictly forbid it. |
| 18:27.25 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 18:27.31 | [TK]D-Fender | </sarcasm> |
| 18:27.53 | Navion | <== I'm not worthy... |
| 18:28.47 | [TK]D-Fender | Navion: First step is admitting you have a problem... |
| 18:28.55 | *** join/#asterisk macros73 (n=cs@dsl093-063-226.pit1.dsl.speakeasy.net) |
| 18:28.57 | Navion | I just hate to manually manipulate database entries not knowing how good the locks are and what else is trying to manipulate the db at the same time |
| 18:29.08 | Navion | Done that... |
| 18:29.58 | [hC] | If i send a NOTIFY to a polycom phone for it to check its config (and reboot if there is new config available) - what happens if i send this to a phone while someone is on a call? |
| 18:30.11 | [hC] | will it wait til they are off and then do it? or reboot mid-call? |
| 18:30.32 | Qwell | [hC]: it'll pierce the users temple, and reboot |
| 18:30.39 | Qwell | (no, I have no idea) |
| 18:30.44 | errr | when you post a question to the asterisk-users mailing list how long does it normally take before you question shows up on the list? |
| 18:30.45 | [N00B] | [TK]D-Fender: is there anywhere I can see what is being called by format_wav? I cannot for the life of me figure out what is being called. here is the other thing... the issue just started happening in the last few minutes. We have not made a change to the system since the 21st of the month. which was just a name change. and from then it has been working all year. |
| 18:30.56 | rpm | [hC], it will wait until the user is off the phone. it does a graceful restart. |
| 18:30.56 | [hC] | Qwell: damn, i was abotu to hit enter too. for this user, i would be thrilled. |
| 18:31.07 | [hC] | rpm: great. thanks :) |
| 18:32.13 | *** join/#asterisk Strom_M (n=strom@m5e0e36d0.tmodns.net) |
| 18:32.29 | [hC] | rpm: you use 2.2.0... have you set anyones callwaiting type to "ring" instead of "beep" in the config? Seems my vanishing ringer volume is related to that |
| 18:33.05 | *** join/#asterisk alayho (n=kevin@12.40.200.74) |
| 18:33.36 | rpm | [hC], nope. and actually i was wrong.. we are still running 2.0.2 :) |
| 18:33.44 | rpm | misplaced the digits. |
| 18:34.08 | [hC] | Ahh.. |
| 18:34.09 | [hC] | gotcha. |
| 18:34.36 | ManxPower | errr: from 5 mins to 48 hours |
| 18:35.02 | errr | ManxPower: ah, so I guess Ill just sit back and wait longer :) thanks |
| 18:36.44 | *** part/#asterisk [N00B] (n=ckwall@206.71.78.172) |
| 18:37.49 | ManxPower | errr: most of the time it is less than 2 hours, but don't count on it. The mailing list gets blasted with spam frequently and that causes the machines that do the spam filtering to slow down |
| 18:40.58 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 18:40.59 | coolfreecode | howhey |
| 18:41.23 | coolfreecode | hey guys who knows a tutorial to create a trunk iax |
| 18:41.39 | coolfreecode | plz |
| 18:42.06 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:42.09 | [TK]D-Fender | ~jerjerguide |
| 18:42.09 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 18:42.30 | [TK]D-Fender | nope... sip only |
| 18:42.38 | [TK]D-Fender | coolfreecode: your ITSP should have a sample |
| 18:43.57 | coolfreecode | i want join 2 asterisk with a trunk iax |
| 18:45.37 | _Sam-- | do it |
| 18:46.13 | _Sam-- | set them both up, then have one machine register via IAX to the other. |
| 18:46.18 | _Sam-- | not that much to it |
| 18:46.21 | coolfreecode | how to connect Asterisk to Asterisk using IAX2 Trunk |
| 18:46.35 | _Sam-- | iax.conf ? |
| 18:47.12 | *** join/#asterisk Strom_C (n=strom@m240e36d0.tmodns.net) |
| 18:47.32 | coolfreecode | [servidora] |
| 18:47.32 | coolfreecode | type=friend |
| 18:47.32 | coolfreecode | username=servidorb |
| 18:47.32 | coolfreecode | secret=password |
| 18:47.32 | coolfreecode | auth=plaintext |
| 18:47.32 | coolfreecode | host=dynamic |
| 18:47.34 | coolfreecode | peercontext=entrantes |
| 18:47.36 | coolfreecode | context=entrantes |
| 18:47.38 | coolfreecode | trunk=yes |
| 18:47.48 | coolfreecode | iax.conf server B: |
| 18:48.06 | _Sam-- | ; We can register with another IAX server to let him know where we are |
| 18:48.06 | _Sam-- | ; in case we have a dynamic IP address for example |
| 18:48.06 | _Sam-- | ; |
| 18:48.06 | coolfreecode | [servidorb] |
| 18:48.06 | coolfreecode | type=friend |
| 18:48.06 | coolfreecode | username=servidora |
| 18:48.06 | coolfreecode | secret=password |
| 18:48.06 | coolfreecode | auth=plaintext |
| 18:48.07 | coolfreecode | host=dynamic |
| 18:48.09 | coolfreecode | peercontext=entrantes |
| 18:48.11 | coolfreecode | context=entrantes |
| 18:48.15 | coolfreecode | trunk=yes |
| 18:48.22 | _Sam-- | the contexts are important, but you need to register one machine, to another. |
| 18:49.10 | _Sam-- | maybe you dont have to, depending on what you want to do i guess. i think, you will probably want one to register to another. |
| 18:49.45 | coolfreecode | okas |
| 18:51.00 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
| 18:52.29 | blitzrage | ~pb |
| 18:52.30 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:59.38 | *** join/#asterisk Darthclue (n=root@li13-84.members.linode.com) |
| 19:01.02 | polerin | feh, I'm feeing retarded, but why is Dial(${USER}) not playing audio to the calling party? |
| 19:01.29 | polerin | I thought that you'd hear ringing unless r was specified |
| 19:04.29 | _Sam-- | [TK]D-Fender : you here? |
| 19:04.39 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:04.40 | _Sam-- | our outgoing dtmf no work anymore |
| 19:05.05 | [TK]D-Fender | _Sam--: ? |
| 19:05.25 | _Sam-- | when our employees make outgoing calls that require dtmf input, it doesnt work. |
| 19:05.25 | [TK]D-Fender | _Sam--: Should have nothingt o do with... |
| 19:05.39 | _Sam-- | like if they need to dial '100' for osmeone's extension...when they dial 100, theother side doesnt hear it. |
| 19:05.54 | _Sam-- | i just confirmed. |
| 19:06.11 | _Sam-- | nothing in our settings has changed. |
| 19:06.39 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 19:07.10 | [TK]D-Fender | _Sam--: PM |
| 19:09.20 | *** join/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 19:20.25 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 19:26.41 | coolfreecode | hey |
| 19:26.49 | coolfreecode | what that mean : chan_iax2.c:1968 iax2_destroy: Avoiding IAX destroy deadlock |
| 19:26.55 | coolfreecode | thanks |
| 19:29.29 | *** join/#asterisk bantu (n=Miranda@p54A33653.dip0.t-ipconnect.de) |
| 19:30.36 | *** join/#asterisk CVirus (n=GoD@62.135.96.14) |
| 19:31.56 | *** join/#asterisk Strom_M (n=strom@m340e36d0.tmodns.net) |
| 19:32.32 | coolfreecode | hey guy's what that mean : chan_iax2.c:1968 iax2_destroy: Avoiding IAX destroy deadlock |
| 19:35.32 | *** join/#asterisk gardo (n=gardo@61.14.191.140) |
| 19:36.34 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:38.03 | bkruse | coolfreecode: you using queues? |
| 19:38.03 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:38.34 | coolfreecode | no queue yes trunk iax |
| 19:39.08 | *** join/#asterisk CVirus (n=GoD@62.135.96.14) |
| 19:39.16 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:40.55 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 19:42.29 | jsaunders | Anyone have an idea when the Switchvox/Asterisk Business Edition integration will be finalized and available to public? |
| 19:46.24 | russellb | jsaunders: i'm not exactly sure where you got that from, but there isn't an answer to the question |
| 19:46.42 | russellb | i can say that you can expect the same quality level of support for switchvox as a digium product, as you would for be |
| 19:48.47 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 19:49.31 | Strom_M | russellb: apparently someone has been turning the official digium rumor crank again |
| 19:49.38 | jsaunders | russellb: I was told my customer service that in the new year I will be able to upgrade our Business Edition copy to one based off Switchvox. |
| 19:49.49 | jsaunders | For free, based off our current license. |
| 19:50.07 | Qwell | jsaunders: that's quite different from being integrated |
| 19:50.22 | *** join/#asterisk Arno[Slack] (n=hellSOUN@gre92-1-81-57-177-108.fbx.proxad.net) |
| 19:50.52 | jsaunders | True. I would think the company would use both for one super product, thus my use of the word integrate. |
| 19:51.52 | russellb | business edition and switchvox are _very_ different products |
| 19:52.21 | russellb | it may be the case that your situation would be better suited to switchvox, but that's not necessarily the case |
| 19:52.56 | [TK]D-Fender | yup, tahts yeah, thats like comparing apples to... crabapples |
| 19:53.42 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 19:53.57 | hmmhesays | well I think I found the answer to my m22 problem |
| 19:55.15 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 19:55.45 | Nugget | so, I'm overdue for a nice overhaul of my dialplan. the company's grown quite a bit since I wrote it and there's a lot of cruft and vestigial code in it. |
| 19:56.01 | Nugget | is it time to look at ael or whatever, or should I just stick with the old-and-busted syntax? |
| 19:56.40 | Nugget | hold tight for lua in 1.6? :) |
| 19:57.38 | Nivex | lua? |
| 19:58.17 | Nugget | http://lua.org/ |
| 19:58.42 | Nugget | a great move for asterisk for anyone who values code portability between their pbx and their world of warcraft addons. |
| 19:59.00 | russellb | lol |
| 20:01.06 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 20:01.14 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 20:02.20 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-187.pskn.east.verizon.net) |
| 20:03.21 | Nivex | neat little language. were you joking about lua in 1.6, or is that on the timeline? |
| 20:03.54 | Nugget | someone wrote it. dunno if the community will embrace it |
| 20:04.16 | Nugget | http://www.russellbryant.net/blog/?p=20 <-- some guy named russell blogged about it here :) |
| 20:05.25 | russellb | o.O |
| 20:05.30 | russellb | imposter! |
| 20:05.40 | Nugget | heh |
| 20:05.42 | niekie | :o |
| 20:06.29 | MrChimpy | so has ael been deemed as crap as dialplan? |
| 20:06.43 | russellb | no, it has not |
| 20:06.49 | Nugget | ael or ael2? :) |
| 20:06.53 | russellb | it is being very widely used, and still actively developed |
| 20:06.57 | russellb | ael in 1.4, that is |
| 20:07.07 | Nugget | I really don't grok the implications and I'm hesitant to use it for no reason I can adequately articulate |
| 20:07.09 | MrChimpy | ok, so it's just chucking another language at the problem |
| 20:07.37 | MrChimpy | I await the javascript, java, perl, C etc dialplan interfaces with interest :) |
| 20:07.39 | russellb | Nugget: friendlier syntax .... much easier to write complex logic |
| 20:07.45 | russellb | having loops and conditional statements and stuff |
| 20:07.51 | [TK]D-Fender | s/widely/sparsely |
| 20:07.58 | russellb | [TK]D-Fender: lies |
| 20:08.05 | [TK]D-Fender | TRUTH |
| 20:08.06 | Nugget | hold out for INTERCAL-DIALPLAN |
| 20:08.15 | russellb | [TK]D-Fender: *shrugs* |
| 20:08.20 | MrChimpy | lolplan! |
| 20:08.24 | Nugget | haha |
| 20:08.30 | russellb | i can haz callerid(num) ? |
| 20:08.47 | Corydon76-dig | I think it's been widely tried, but I don't know what the extent is for its usage |
| 20:09.00 | Nugget | I'M IN U'R IVR DIALIN' U'R EXTENSIONS |
| 20:09.32 | *** join/#asterisk j0wbl4ck (n=jowblack@201.78.22.62) |
| 20:09.59 | j0wbl4ck | hello guys |
| 20:10.39 | j0wbl4ck | i need help, you are know any providers SIP, or providers reseller plains in voip? |
| 20:10.39 | MrChimpy | can we call extensions buckits? |
| 20:10.46 | j0wbl4ck | [Corydon76-dig]: SIP for Asterisk? |
| 20:11.01 | j0wbl4ck | SIP for asterisk? |
| 20:13.13 | j0wbl4ck | anybody know? |
| 20:14.29 | j0wbl4ck | ><> |
| 20:15.09 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 20:16.19 | [TK]D-Fender | j0wbl4ck: Try putting that question into an other that actually makes sense. |
| 20:16.29 | *** join/#asterisk Strom_C (n=strom@m700e36d0.tmodns.net) |
| 20:22.41 | [TK]D-Fender | order* |
| 20:30.54 | j0wbl4ck | ok |
| 20:34.48 | Navion | Anyone help with speaker paging. console/dsp device debugging and such? |
| 20:38.57 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:39.44 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 20:46.21 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:46.34 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 20:46.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:46.37 | _Sam-- | [TK]D-Fender : another sitchu. im sure you have an answer, as always. our incoming call queues are not listening to the ring "strategy" |
| 20:46.42 | *** join/#asterisk saftsack (n=saftsack@pD9E07A55.dip.t-dialin.net) |
| 20:46.48 | _Sam-- | my strategy is "ring all" |
| 20:46.57 | _Sam-- | but the employees are saying only one phone at a time is ringing |
| 20:47.07 | _Sam-- | im not seeing that, on this queue, though |
| 20:48.41 | *** join/#asterisk gpowers (n=gpowers@208.66.168.244) |
| 20:52.15 | *** part/#asterisk gpowers (n=gpowers@208.66.168.244) |
| 20:53.55 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 20:54.25 | *** join/#asterisk Violater (n=vioman@d193-48-24.home3.cgocable.net) |
| 20:55.05 | Violater | anyone good with openser and asterisk please msg me quick question |
| 20:56.06 | jameswf | ~openser |
| 20:56.07 | jbot | openser is, like, an open source GPL project that aims to develop a robust and scalable SIP server. It is spawned from FhG FOKUS SIP Express Router (SER) and it promotes a development strategy open for contributors and contributions. From project's website http://www.voip-info.org/wiki/view/About+OpenSER |
| 20:56.47 | Violater | a technical question |
| 20:57.30 | jameswf | jbot: tell Violater ask |
| 20:57.50 | *** join/#asterisk barhom (n=barhom@h-89-233-192-113.wholesale.rp80.se) |
| 20:57.50 | jameswf | jbot: ask | Violater |
| 20:57.56 | jameswf | stupid bot |
| 20:57.58 | j0wbl4ck | msn |
| 20:58.35 | wwalker | ~ask |
| 20:58.35 | 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. |
| 20:58.47 | barhom | Hi, I have a question about changing the language=en to language=se, whatever I do I keep hearing english stuff, what am I doing wrong? If I use Set(LANGUAGE()=se) in extentions this is working fine |
| 20:59.44 | barhom | I am editing the language=se in sip.conf |
| 20:59.58 | Violater | ok i have openser running on a linksys wrt54gl with 2 aastra 55i phones behind it with their outbound proxy set to the router ip and registrar and proxy ip set to my external asterisk 1.4.14 server, both phones set to canrevite=yes and the provider for my did as well.. everything works perfect except when i put a call on hold and take it off there's one way audio |
| 21:01.46 | jameswf | sounds like a poop in the natting |
| 21:03.20 | Violater | i'm pretty sure its a rtpproxy/openser configuration issue i just don't know what |
| 21:03.46 | Violater | because internal calls and transfers and inbound outbound all work perfect audio wise |
| 21:07.26 | barhom | where is the correct place to put wide-use language=xx for my sip configuration? |
| 21:09.14 | nohup_ | is it very unusual to have "Channel: SIP/some_number@some_provider" in a call-file ? |
| 21:10.02 | nohup_ | cause... it did work as long as i put a local SIP phone there... but when it's an external one... it _sometimes_ does ring it... (1 out of 10 times, or something), but there's no audio |
| 21:10.36 | nohup_ | (while calling that number through SIP directly works fine) |
| 21:12.12 | JT | Violater: canreinvite=no |
| 21:13.53 | *** join/#asterisk robeph (n=robf@24.214.206.254) |
| 21:15.03 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 21:15.36 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
| 21:15.55 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 21:19.21 | *** join/#asterisk BadHorsie (n=illidan@ip254-10.ct.co.cr) |
| 21:20.26 | *** join/#asterisk philth (n=philth@d38-181-68.home1.cgocable.net) |
| 21:21.54 | philth | I have a sip # call an extention. Once it is there I want it to wait for another number. Then dial out to whatever number is dialed. This should be easy, but I cant seem to get it. |
| 21:23.29 | *** join/#asterisk mascool (n=george@adsl-76-226-150-178.dsl.sfldmi.sbcglobal.net) |
| 21:23.58 | mascool | does anyone why a polycom ip501 would have the mwi led blinking while there's no message in the mailbox ? |
| 21:24.27 | _Sam-- | philth : background / digit time out / response timeout / dial ext |
| 21:24.31 | nestAr | mascool: it's checking the wrong mailbox |
| 21:25.13 | mascool | i triple checked sip.conf and it's not checking the wrong mailbox |
| 21:25.24 | nestAr | i guess it just hates you. |
| 21:25.30 | mascool | no shit |
| 21:25.31 | mascool | :) |
| 21:25.34 | CrazyTux | Polycoms are simply evil. |
| 21:25.50 | mascool | but all 4 of them ? |
| 21:26.01 | jer | anybody have any experience with asterisk on freebsd? just wondering how well it works before i set up my new box |
| 21:26.10 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 21:26.22 | yangvnc | I have the following line in extensions exten => 2968,1,Dial(SIP/2968&SIP/2931) it rings on 2 numbers. Now I am wondering how I could ring 2968 and a telephone number 00<countrycode><number> ? |
| 21:27.04 | barhom | Is it an fxs card I need if Im looking to do this, SIP Provider->asterix->my normal phones @ home ? |
| 21:27.26 | JT | asterisk |
| 21:27.30 | JT | not asterix |
| 21:27.44 | [TK]D-Fender | barhom: it would be, but I highly advise using an ATA like the Linksys SPA-2102 for things like that instead. Far cheaper and more flexible |
| 21:27.44 | barhom | asterisk* I keep getting that wrong, but focus on the question please |
| 21:28.01 | barhom | I have an UTSTARCOM ian-02ex ata box |
| 21:28.10 | barhom | though I dont have the login/pass to it cuz my old provider locked it |
| 21:28.27 | [TK]D-Fender | barhom: then forget about it |
| 21:28.34 | JT | don't tell us to focus on the question if you can't even focus on getting the name right :) |
| 21:29.02 | barhom | [TK]D-Fender: forget about it as, there is no way to come in to the box ? |
| 21:29.13 | _Sam-- | the utstarcomm default passwords are 8888 i think |
| 21:29.17 | _Sam-- | for the f1000 anyway |
| 21:29.22 | [TK]D-Fender | barhom: I'm saying forget about your locked ATA. Locked is locked |
| 21:29.25 | JT | yes you need an fxs port |
| 21:29.48 | [TK]D-Fender | barhom: So go buy a normal unlocked one like the Linksys I referred you to. |
| 21:29.57 | barhom | Ill look into it fender |
| 21:30.12 | barhom | does callerid work with ata box? |
| 21:30.20 | _Sam-- | i bought a locked utstarcomm wifi phone, and unlocked it fine. |
| 21:30.22 | CrazyTux | [TK]D-Fender, have much experience with AMI / Manager.conf stuff? |
| 21:30.30 | [TK]D-Fender | CrazyTux: A little |
| 21:30.36 | [TK]D-Fender | barhom: Yup |
| 21:30.55 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 21:31.03 | ZX81 | ~seen critch |
| 21:31.06 | jbot | critch <n=critch@c-71-228-211-57.hsd1.tn.comcast.net> was last seen on IRC in channel #asterisk, 50d 11m 21s ago, saying: 'isn't the 1-4 branch essentially a release that I just happened to use svn to download?'. |
| 21:31.48 | barhom | alright thanks a lot for the help guys, I guess Ill try to unlock my utstarcom before I go buy another ata box, but anyone that can help with the "language=se" problem I have? Ive tried putting language=se like everywhere in all the files even if I dont use them, it just simply aint working |
| 21:31.52 | jameswf | ~seen jameswf-HOME |
| 21:31.53 | jbot | jameswf-home <n=that@ip72-204-228-104.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 2d 14h 40m 32s ago, saying: '~bye'. |
| 21:32.02 | barhom | Set(LANGUAGE()=se) works though, but I dont want to put that for every extension |
| 21:32.03 | _Sam-- | what is the utstarcomm model number you have |
| 21:32.04 | jameswf | NEAT |
| 21:32.12 | barhom | _Sam--: it is utstarcom ian-02ex |
| 21:33.07 | JT | barhom: what nationalisation do you actually need? |
| 21:33.13 | barhom | se JT |
| 21:33.18 | barhom | I have all the sounds |
| 21:33.24 | _Sam-- | barhom : your ATA DOES have a default password. |
| 21:33.29 | _Sam-- | if you can find out what it is, you can get it. |
| 21:33.38 | JT | so it's the prompts you are worries about, barhom ? |
| 21:33.51 | barhom | jt: Im not really understanding what you mean |
| 21:34.01 | barhom | I want when I Playback(invalid) it plays from se/invalid.gsm |
| 21:34.13 | JT | barhom: only a few things are affected by setting a language |
| 21:34.13 | barhom | I read in the docu that you set "language=se" in [general] for this |
| 21:34.18 | JT | that's a prompt |
| 21:34.23 | JT | an audio prompt |
| 21:34.24 | barhom | okay, then a prompt yes |
| 21:34.26 | _Sam-- | barhom: this will help you get into the ATA, i think: http://forum.sipphone.com/viewtopic.php?t=1798&highlight= |
| 21:34.33 | barhom | thanks sam |
| 21:34.36 | JT | have you set up indications.conf? |
| 21:34.44 | barhom | no, I havent |
| 21:34.47 | ZX81 | anyone know if there is an irc room for misdn? |
| 21:34.51 | JT | then you might want to do that |
| 21:35.05 | barhom | country=us ; default location |
| 21:35.07 | nohup_ | hmmm |
| 21:35.08 | JT | ZX81: #pain-and-suffering ;) |
| 21:35.09 | barhom | change to to se? |
| 21:35.16 | nohup_ | does anyone know of a command-line SIP client ? |
| 21:35.16 | JT | barhom: yes |
| 21:35.17 | ZX81 | JT: not wrong! :) |
| 21:35.27 | JT | ZX81: what problems are you having? |
| 21:35.41 | JT | nohup_: sipp is cli based for testing sip |
| 21:35.44 | nohup_ | i setup call files, but it won't let me use a channel that dais out on SIP |
| 21:35.49 | nohup_ | so i need an alternative :) |
| 21:35.57 | nohup_ | sipp... okay... |
| 21:36.01 | ZX81 | trying to get octasic echo can working with misdn but it complains I need to recompile Asterisk because I have 1.4+ |
| 21:36.04 | ZX81 | so I do |
| 21:36.07 | ZX81 | but it says the same again |
| 21:36.08 | ZX81 | :) |
| 21:36.13 | ZX81 | will pastebin it |
| 21:36.14 | ZX81 | :) |
| 21:36.18 | JT | hmm |
| 21:36.29 | JT | i just try and avoid misdn |
| 21:36.33 | JT | it's renamed isdn4linux hell |
| 21:36.52 | ZX81 | yep |
| 21:37.11 | ZX81 | I was trying to support digium - little did I know its basically a beronet thingy |
| 21:37.12 | ZX81 | http://pastebin.ca/799780 |
| 21:38.44 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:38.57 | barhom | didnt work JT, I have set the indictions, country=se.. and also language=se under [general] in sip.conf it still plays the english audio files this is disturbing because Ive read so many docs on how to change language on google but none of them work |
| 21:39.15 | ZX81 | where are your se files? |
| 21:39.32 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:39.32 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 21:40.14 | barhom | /var/lib/asterisk/sounds/se/ and /usr/share/asterisk/sounds/se/ |
| 21:40.20 | barhom | they are symlinked |
| 21:40.20 | JT | barhom: did you restart asterisk? |
| 21:40.24 | barhom | yes JT |
| 21:40.33 | deeperror | is there a way for inbound call to goto a queue then be answered by an agent that dials a specified feature code? Like feature code directs first in queue to station dialing code? |
| 21:40.45 | barhom | I know the files for se are working because if I do "Set(LANGUAGE()=se" it works |
| 21:41.03 | deeperror | or what would something like this be called if it has a name |
| 21:42.02 | ZX81 | deeperror sounds like you just want to ring somewhere and then do a pickup(bla) |
| 21:42.31 | ZX81 | type show application Pickup |
| 21:42.44 | ZX81 | or core show application pickup |
| 21:42.51 | deeperror | on it |
| 21:43.46 | ZX81 | kinda weird |
| 21:43.50 | deeperror | so the channel would have to stay in state ringing? Wouldn't this prevent the inbound from hold music etc? |
| 21:43.55 | ZX81 | cos it will need to be ringing somewhere |
| 21:44.06 | ZX81 | nah you can have hold music |
| 21:44.09 | ZX81 | on a ringing call |
| 21:44.16 | ZX81 | dial(bla|30|m) |
| 21:44.32 | ZX81 | might want to dial some local thingy |
| 21:44.34 | ZX81 | like |
| 21:44.49 | ZX81 | dial(Local/123@test||m) |
| 21:44.53 | deeperror | my setup is agents that do inbound and outbound dialing...they have analog phones for the moment and we need to use those so I was going to setup an extension that just is indicator lights when the lights are blinking they could dial an extension and connect with the first person in line in the queue |
| 21:45.26 | ZX81 | yeah if you had it calling some indicator extension |
| 21:45.29 | ZX81 | that doesn't answer |
| 21:45.36 | ZX81 | they could dial 123 or whatever |
| 21:45.38 | ZX81 | to run pickup |
| 21:45.48 | deeperror | then logic in the context the agents default to have the pickup application in there to grab them off the call |
| 21:46.06 | deeperror | ok i'll check into this more |
| 21:46.09 | ZX81 | you need to set pickupmark thing |
| 21:46.17 | ZX81 | depends on which version of asterisk you're using |
| 21:46.42 | deeperror | seems like an option for sure its just our limitations here i'm trying to work around and 25 bucks for a light indicator is cheaper than new phones for everyone haha |
| 21:46.55 | ZX81 | :D |
| 21:46.59 | ZX81 | heh always the way |
| 21:47.49 | deeperror | also the ringing is heard in recordings and call waiting beeps are picked up on the oldschool recording they use here so it's always a hack ha |
| 21:49.20 | jameswf | anyone woh knows of a did provider for iraq/afganistan or the general area contact wvroger |
| 21:51.03 | *** part/#asterisk JayTee52 (n=jforde05@207-67-84-185.static.twtelecom.net) |
| 21:52.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:52.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:56.48 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-235.dsl.peoril.sbcglobal.net) |
| 21:57.24 | MACscr | Anyone know where i can get a DID for istanbul, turkey? |
| 21:58.07 | MACscr | I thought voxbone had them (which they list on their site), but when you go to to order them, they dont have them. =( |
| 21:58.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:00.13 | *** join/#asterisk BadHorsie (n=illidan@ip254-10.ct.co.cr) |
| 22:00.50 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.12.183) |
| 22:01.30 | *** join/#asterisk saftsack (n=saftsack@pD9E07A55.dip.t-dialin.net) |
| 22:03.10 | jozu | When purchasing a DID voxbone, howto setup in asterisk? |
| 22:08.30 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-0-23.red.bezeqint.net) |
| 22:08.57 | Dovid | what codec/protocl does HD Voice use (for instance on the Polycom 650) |
| 22:09.00 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 22:09.27 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b206d38.dorm.bilkent.edu.tr) |
| 22:11.24 | mcab | Dovid: G722, IIRC |
| 22:13.02 | *** part/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 22:13.53 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 22:14.17 | Dovid | mcab: is that OpenSource or closed ? |
| 22:14.25 | Dovid | nm |
| 22:14.26 | Dovid | will gogole |
| 22:15.27 | JT | open source |
| 22:16.08 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 22:18.00 | Dovid | JT: And i assume asterisk supports it ? |
| 22:18.35 | JT | in trunk i think |
| 22:18.56 | JT | the asterisk core was only designed for 8kHz audio |
| 22:19.18 | JT | so it was a bit of work to get stuff like this working |
| 22:19.45 | russellb | 1.4 supports g722 as well, to some degree |
| 22:19.51 | russellb | passthrough, playback, and recording |
| 22:20.08 | russellb | though it is trivial to backport codec_g722 from trunk for transcoding |
| 22:21.08 | *** join/#asterisk Strom_M (n=strom@m670e36d0.tmodns.net) |
| 22:22.19 | Dovid | russelb: what do u mean by to a degree i it supports playbak, recording and passthrough ? |
| 22:24.59 | russellb | it supports those in 1.4 natively |
| 22:25.02 | russellb | it does not transcode in 1.4 natively |
| 22:25.05 | *** join/#asterisk BadHorsie (n=sebas@ip254-10.ct.co.cr) |
| 22:26.57 | Dovid | russellb: So if the phones want to use g722 in 1.4 it will only work if the phone it is calling supports g722 |
| 22:28.02 | russellb | correct |
| 22:28.16 | [TK]D-Fender | G.722 = whatever |
| 22:28.25 | Qwell | = whatever? |
| 22:28.28 | russellb | you're ... whatever |
| 22:30.41 | twisted | hah |
| 22:30.48 | barhom | anyone know any freeware that can record in .gsm codec to make new playback files? |
| 22:30.51 | barhom | for windows. |
| 22:31.10 | twisted | barhom: you can use wav files too |
| 22:31.19 | twisted | 8khz mono |
| 22:31.23 | barhom | k thanks twisted |
| 22:31.44 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:32.55 | philth | Are WaitExten and Background interchangeable |
| 22:33.17 | philth | I use Background and it works, but WaitExten doesn't do Anything, just times out. |
| 22:34.21 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:34.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:37.29 | *** join/#asterisk Belgarath (i=belgarat@banda.pl) |
| 22:37.35 | Belgarath | hello |
| 22:38.15 | Belgarath | does anybody have a linksys WRTP54G ? |
| 22:38.25 | Belgarath | i have a strange problem with it |
| 22:38.40 | Belgarath | it is connected to my asterisk box |
| 22:38.54 | Belgarath | and all calls going outside and echo on my asterisk box are ok |
| 22:39.31 | Belgarath | bvut when i call any other number siting on my asterisk box (doing anything else than echo or ringing) there is no voice in my headphone |
| 22:49.17 | *** join/#asterisk Strom_C (n=strom@m560e36d0.tmodns.net) |
| 22:50.50 | *** join/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 23:01.51 | Navion | Anyone have loud speaker paging through the PC audio port working? |
| 23:02.13 | [TK]D-Fender | Navion, Dial(OSS/dsp) <- |
| 23:02.46 | Navion | Yes... |
| 23:02.49 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 23:03.26 | Navion | OSS/dsp not console/dsp? |
| 23:04.31 | [TK]D-Fender | Navion, correct |
| 23:04.39 | Navion | I have the following in extensions_custom.conf |
| 23:04.45 | Navion | exten => *51,1,Dial(console/dsp) |
| 23:04.46 | Navion | exten => *51,2,Playback(custom/bosun) |
| 23:04.46 | Navion | exten => *51,3,Hangup() |
| 23:05.08 | Navion | Should be OSS? |
| 23:05.43 | Navion | Is it OSS/bosun for the playback of the alert tone too? |
| 23:06.15 | Qwell | ~freepbx |
| 23:06.16 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:06.45 | Navion | OSS in caps or lower case? The channel was set up as chan_oss |
| 23:07.02 | *** part/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 23:08.38 | iamthelostboy | if i want any user to be able to pickup an incoming call on another phone, what do i need to setup? "answer groups" is in my head.. but doesn't sound right, and google didnt do well by me when searching for it |
| 23:08.53 | barhom | it this the correct way to add "08" infront of all numbers that does not start with 0, "exten => _[123456789].,1,Dial(SIP/08${EXTEN}@digisip) |
| 23:09.16 | *** part/#asterisk Freman (n=freman@brdr-gw-01.benon.com) |
| 23:09.37 | Navion | ~ring group |
| 23:09.38 | jbot | before group dies, they see the ring |
| 23:09.44 | *** join/#asterisk saftsack (n=saftsack@pD9E07A55.dip.t-dialin.net) |
| 23:09.50 | Navion | ~ringgroup |
| 23:10.37 | Navion | iamthelostboy: Do you want all the extensions to ring? If so, set up a ring group and include all the extensions then anyone can pick it up. |
| 23:11.25 | Navion | ~call pickup |
| 23:11.26 | jbot | ACTION looks around and then screams out pickup as loudly as possible |
| 23:11.43 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 23:12.48 | iamthelostboy | no, i just want a single extension to ring, and then other phones to be able to pick it up... callgroup= and then *8# or *8 sounds like it is what im after |
| 23:12.51 | iamthelostboy | *trys* |
| 23:13.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:19.18 | iamthelostboy | eep! that didnt go well |
| 23:20.02 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 23:20.11 | jozu | i have a dude, voxbone can giveme a DID number for incoming calls.. example 9XXXXX.. but howto configure the external calls from my sips with the same DID 9XXXXX????? |
| 23:21.57 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 23:21.58 | _ShrikE | Set(CALLERID(num)) |
| 23:25.05 | jozu | But if my provider voip not let me? |
| 23:25.51 | nestAr | then you need a better provider |
| 23:26.40 | jozu | Thanks for the answers |
| 23:27.31 | nestAr | most voip providers will let you set your caller id to at least the numbers they've assigned you |
| 23:27.42 | nestAr | some of them will let you set your caller id to whatever you want. |
| 23:27.57 | barhom | how do I parse this dial command correctly to put "08" infront of the number dialed: Dial(SIP/08${EXTEN}@digisip) |
| 23:28.06 | barhom | that there isnt working atm |
| 23:28.52 | nestAr | that should work. |
| 23:28.52 | _ShrikE | barhorn: get rid if the SIP/ |
| 23:28.56 | nestAr | exten => _NXXXXXX,4,Dial(SIP/1502${EXTEN}@vitel-outbound) ; 7 digit dialing hack |
| 23:29.10 | nestAr | is what i use |
| 23:29.11 | _ShrikE | maybe not |
| 23:29.12 | *** join/#asterisk saftsack (n=saftsack@pD9E07A55.dip.t-dialin.net) |
| 23:29.39 | barhom | hmm, I dont need the SIP/ ? |
| 23:29.49 | nestAr | yes, if you're dialing SIP |
| 23:29.50 | perf3kt | ~book |
| 23:29.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 23:29.51 | _ShrikE | sorry. you do |
| 23:31.21 | jameswf | ~hack |
| 23:33.20 | jozu | nestAr in this line exten=s,1,Dial(${ARG1}) can use 9XXXX,1,Dial(${ARG1}) ? |
| 23:33.53 | nestAr | jozu: i am not sure what you're asking. |
| 23:34.44 | nestAr | ARG1 would be from a macro |
| 23:34.55 | nestAr | so that syntax isn't 100% |
| 23:35.22 | nestAr | something like "exten => _9XXXX,1,Dial(${EXTEN}) would work though. |
| 23:35.49 | jozu | I will try |
| 23:36.15 | blitzrage | nestAr: you forgot to supply a technology |
| 23:36.17 | nestAr | you'll probably need something else in the Dial command |
| 23:36.24 | blitzrage | SIP/${EXTEN} |
| 23:36.28 | nestAr | Sip, IAX, etc |
| 23:37.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:39.13 | barhom | ${EXTEN} isnt that only the number that you dialed? for example if I change this; Dial(SIP/${EXTEN}@digisip) |
| 23:39.19 | barhom | to Dial(SIP/0856166300@digisip) |
| 23:39.38 | barhom | shouldnt that always dial that 0856166300 number, whatever I type (aslong as it matches it ofcourse) |
| 23:39.50 | nestAr | sure |
| 23:40.00 | barhom | hmm, yeh, I thought so as well but its not dialing it.. |
| 23:40.11 | nestAr | depending on what the exten => X, is |
| 23:40.17 | nestAr | where X is your patern |
| 23:40.21 | barhom | yeh ofcourse nestAr |
| 23:40.29 | barhom | exten => _[123456789].,1,Dial(SIP/0856166300@digisip) |
| 23:40.38 | barhom | and I type in "5616630" |
| 23:40.45 | barhom | it should call 0856166300 |
| 23:40.48 | Strom_C | you know, you can just use Z to match digits 1-9 |
| 23:40.59 | barhom | great Strom_C ;) |
| 23:41.11 | barhom | so _Z., ? |
| 23:41.17 | Strom_C | yes |
| 23:41.43 | *** join/#asterisk tzafrir_laptop (i=tzafrir@192.117.42.208.static.012.net.il) |
| 23:43.23 | jozu | i have this: |
| 23:43.24 | jozu | exten=_9XXXXXXXX,1,Macro(trunkdial,${trunk_1}/${EXTEN:0}) |
| 23:43.27 | jozu | in my dialplan |
| 23:44.00 | jozu | in macro-trunkdial |
| 23:44.08 | jozu | exten=s,1,Dial(${ARG1}) |
| 23:44.13 | jozu | first line |
| 23:44.29 | JT | and if not Z |
| 23:44.35 | JT | at least [1-9] |
| 23:45.24 | jozu | howto put my DID number 9xxxx? |
| 23:46.01 | JT | i don't understand the question |
| 23:49.25 | jozu | JT, i need a setcallerid in external calls, With the same number that my DID from voxbone (9xxx) |
| 23:50.20 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:51.04 | Nugget | setcallerid is setcallerid is setcallerid. |
| 23:51.13 | Nugget | either it works or it doesn't, but that's not something you can control. |
| 23:51.28 | Nugget | set it and either it works or ot. |
| 23:51.30 | Nugget | er, not. |
| 23:52.00 | SwK | Nugget, dont forget they have that new function... |
| 23:52.15 | SwK | set(CALLERID(num)=1900909JEFF) |
| 23:52.47 | SwK | and make sure you dont have spaces around the = sign or you'll create a cariable w/ a space and have a leading space in your number |
| 23:53.29 | Nugget | right, but jozu appears to be asking for a "special" set callerid function for "external" calls. |
| 23:53.37 | Nugget | as if the function knows or cares about "external" |
| 23:53.41 | Nugget | (which it does not) |
| 23:53.54 | iamthelostboy | i cant get callgroup working.. i have callgroup=1 under each sip user, ive got pickupexten = *8 in features.conf, ive restarted asterisk, and when i try to pickup, i get a call failed.. any ideas? |
| 23:57.08 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 23:58.19 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |