| 00:01.54 | *** join/#asterisk adyer (n=adyer@adsl-69-223-13-78.dsl.chcgil.ameritech.net) |
| 00:05.15 | ManxPower | Every day is newbie day with Yourname` |
| 00:05.42 | Yourname` | You know how it is, ManxPower ;) |
| 00:05.43 | trnzmeta | guys for zaptel their versioning any relation to asterisk core branch versioning? |
| 00:05.48 | ManxPower | maqr: /path/to/src/asterisk/doc and /path/to/src/asterisk/configs |
| 00:05.58 | ManxPower | Yourname`: No, I actually don't. |
| 00:06.09 | Yourname` | ManxPower: You just said it, didn't you? |
| 00:06.19 | trnzmeta | zaptel-1.2 relates to asterisk 1.2? |
| 00:06.27 | ManxPower | trnzmeta: correct |
| 00:06.28 | Yourname` | Why would you say it, unless you... don't? |
| 00:06.30 | trnzmeta | zaptel-1.4 related to asterisk 1.4 |
| 00:06.41 | trnzmeta | ManxPower: thanks |
| 00:06.45 | ManxPower | Yourname`: I don't understand people in the KKK either, but I know what they do. |
| 00:07.06 | hardwire | drink |
| 00:07.14 | Yourname` | and drink more |
| 00:08.29 | hardwire | reminds me of new harold and kumar movie |
| 00:08.48 | Yourname` | I laughed my ass of at the theatre! |
| 00:11.57 | *** join/#asterisk schreckdebase (n=tpdc@koe67-2-82-238-198-119.fbx.proxad.net) |
| 00:27.15 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:27.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:37.35 | *** join/#asterisk budol (i=budol@202.124.138.72) |
| 00:46.42 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:47.12 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-42feb375841406a2) |
| 01:00.05 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:00.17 | fujin | so uh |
| 01:00.17 | fujin | 52 active channels |
| 01:00.18 | fujin | 7166 active calls |
| 01:00.22 | fujin | How can I deal with this? |
| 01:00.27 | fujin | It looks like someone is tryign to hax me IVR |
| 01:01.07 | russellb | if it's IP, block them ... |
| 01:03.08 | fujin | unfortunatel not - they're coming through my AS5400 PRI<>SIP->Asterisk clustedr |
| 01:03.33 | fujin | oddly enough, I think they're just dead channels - no one is complaining that they are receiving weird phone calls |
| 01:04.32 | *** join/#asterisk gerphimum (n=trekkie@cpe-24-167-100-36.satx.res.rr.com) |
| 01:08.24 | [TK]D-Fender | fujin, what are you doing that lists 7166 calls? |
| 01:08.55 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 01:09.34 | fujin | show channels |
| 01:10.40 | [TK]D-Fender | fujin, seems odd to list mismatched number of channels to calls. |
| 01:10.42 | *** join/#asterisk mistermocha (n=mistermo@adsl-75-40-195-101.dsl.irvnca.sbcglobal.net) |
| 01:10.56 | fujin | mm |
| 01:11.00 | mistermocha | hey all |
| 01:11.23 | fujin | of note, top/htop believes that /usr/sbin/asterisk has 7000 odd forks |
| 01:11.30 | mistermocha | I've got a question about zapata.conf |
| 01:11.37 | fujin | It's like they're not cleaning up for some reason if they even exist |
| 01:12.39 | mistermocha | when describing out a channel or group, there's all sorts of settings to go with... how does the driver know which properties go with a specific channel or group? |
| 01:13.02 | fujin | [TK]D-Fender: any ideas how i can track down and/or kill whatever the hell is causing it? |
| 01:13.09 | fujin | Wish I knew how to drive this as5400 heh |
| 01:13.11 | [TK]D-Fender | fujin, not sure... |
| 01:14.00 | [TK]D-Fender | mistermocha, everything leading to the "channel =>" line applies to it. The next channel declaration inherits everything that do don't override following it. |
| 01:14.26 | *** join/#asterisk BeeBuu (n=beebuu@218.13.98.192) |
| 01:14.52 | mistermocha | okie |
| 01:14.56 | mistermocha | good to know |
| 01:15.43 | fujin | what a pain. I can't kill it in the middle of the day |
| 01:18.19 | fujin | heh, I think Asterisk might be fucking out. There are only 5 active inbound calls on my as5400 |
| 01:18.26 | fujin | THat's not ideal. |
| 01:23.51 | drmessano | HA |
| 01:24.02 | drmessano | Zaptel = DAHDI.. Confirmed |
| 01:24.12 | drmessano | is probably way late on that |
| 01:24.24 | *** join/#asterisk mirrorcolor (n=iunixan@196.218.222.116) |
| 01:24.38 | russellb | drmessano: was just announced ... |
| 01:24.40 | drmessano | Who's your DAHDI? |
| 01:24.50 | drmessano | Ah |
| 01:25.02 | russellb | trademarks are silly |
| 01:25.04 | drmessano | Well, I heard it from some Brussell Ryant guy on Twitter |
| 01:25.09 | drmessano | I think he's from Belguim |
| 01:25.09 | russellb | <3 |
| 01:25.28 | lanning | the FAQ is under lock and key. |
| 01:25.51 | fujin | blagh.. 7690 calls and counting |
| 01:25.57 | fujin | anyone know what I can do to track this down? |
| 01:26.09 | russellb | waves his hand and says "there are no questions" |
| 01:26.35 | fujin | ugh |
| 01:27.01 | lanning | *** This page intentionally left blank *** |
| 01:30.51 | tzafrir | and what I really don't get is that dahdi.com is owned by someone else |
| 01:31.14 | tzafrir | (though seems to be unused) |
| 01:31.55 | Qwell | .org is owned by Digium. |
| 01:32.08 | Qwell | .com doesn't really matter a whole lot for something like this. Consider that we don't own asterisk.org |
| 01:32.15 | Qwell | erm, don't own asterisk.com |
| 01:32.19 | drmessano | Hmm |
| 01:32.32 | drmessano | banana nut loaf and meatballs |
| 01:32.35 | drmessano | Gotta love parked pages |
| 01:32.51 | drmessano | DADHI is the banana nut loaf of telephony? |
| 01:32.56 | drmessano | Warm.. and home-ish? |
| 01:33.20 | drmessano | Errr |
| 01:33.25 | drmessano | DAHDI |
| 01:34.25 | tzafrir | .org and .net . As of today |
| 01:34.26 | tzafrir | .com was registered by someone from Dubai half a year ago |
| 01:34.31 | fujin | Is there anyway to make a macro quiet? |
| 01:34.46 | tzafrir | core set verbose 0 ? |
| 01:34.47 | Qwell | err, wow. today? |
| 01:34.48 | fujin | like I'm usin ga local channel for hotdesking call delivery, and everytime it tries to deliver it spams like 8 lines |
| 01:34.54 | fujin | tzafrir: no, I still want the other output |
| 01:34.58 | fujin | that is why I said make a macro quiet. |
| 01:35.05 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 01:35.39 | russellb | .net and .org were registrered by digium |
| 01:35.55 | russellb | s/registrered/registered/ |
| 01:36.54 | Nugget | Who's Yer DAHDI? |
| 01:37.14 | tzafrir | well, I guess that a name that is difficult to prononce just makes text chats more valueble |
| 01:37.34 | russellb | it's not that hard to pronounce :) |
| 01:37.44 | russellb | Dah-dee |
| 01:37.46 | Nugget | feh -- there are still people who think Linux is pronounced Lie-nucks and that hasn't seemed to hurt the project any. ;) |
| 01:37.50 | tzafrir | If you're from India |
| 01:42.17 | *** join/#asterisk tobias (n=tobias@cpe-071-070-184-192.nc.res.rr.com) |
| 01:43.58 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 01:44.30 | drmessano | DUNDI, DAHDI, sounds like someone spends way too much time eating Saag Paneer at the "Sacred Cow of New Delhi" restaurant |
| 01:45.34 | fujin | Nugget: Isn't it pronounced Lie-nucks? |
| 01:45.50 | fujin | His name is Lie-nus, after all (not sure if he named it after himself) |
| 01:46.12 | fujin | of note, I say luhnucks (I'm from NZ) |
| 01:46.18 | fujin | fush and chuups |
| 01:46.46 | Qwell | Leeeeeenus |
| 01:46.57 | Qwell | is how his name is pronounced.. |
| 01:47.31 | Nugget | It's pronounced Lih-nux, not Lie-nux. |
| 01:48.11 | denon | Qwell: surely you've heard the old .wav of linus saying "hi, my name is Linus Torvalds, and I pronounce Linux .. Linux" |
| 01:48.15 | Nugget | http://youtube.com/watch?v=z2l86jJqyww |
| 01:48.25 | Qwell | denon: yep |
| 01:48.41 | denon | years and years ago, when it took like 30 minutes to download that one wav on your 56k frame relay connection |
| 01:48.56 | Nugget | heh |
| 01:48.57 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 01:49.24 | Nugget | I'll take that .au file over RMS's share the software song any day of the week. |
| 01:53.31 | drmessano | lol |
| 01:55.10 | fujin | there ya go |
| 01:55.12 | fujin | linux is always linux. |
| 01:55.27 | outtolunc | even on tuesdays |
| 01:57.33 | outtolunc | should have said ... 'unless there is a GPL violation' <G> |
| 02:06.37 | trnzmeta | chan_zap.c: Unable to specify channel 1: No such device or address |
| 02:06.56 | trnzmeta | zaptel starts fine, but asterisk complains |
| 02:07.11 | trnzmeta | any thoughts |
| 02:08.55 | Qwell | trnzmeta: what versions of each, and is it configured? |
| 02:09.07 | Qwell | where it=zaptel.conf |
| 02:09.56 | trnzmeta | both 1.2 |
| 02:10.23 | *** join/#asterisk mike-ekim (n=uideqzer@24.238.28.248) |
| 02:10.36 | *** join/#asterisk TrentCreek (n=trent@cpe-70-116-111-122.rgv.res.rr.com) |
| 02:12.53 | mike-ekim | anyone knwo good links to learn a bit about voip & video? like known codecs, implementation procedures, etc. |
| 02:13.08 | trnzmeta | Qwell: zaptel conf in there |
| 02:13.11 | mike-ekim | something that statys within the realm of asterisk technologies though, or any future known technologies to come to asterisk. |
| 02:14.34 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 02:14.39 | fujin | hi - is there any reason to have /var/spool/asterisk/voicemail when using ODBC storage AND configuration for Voicemail? |
| 02:14.49 | fujin | I notice there are a number of files in there - like it writes out to them before putting them in the db |
| 02:19.22 | fujin | anyone? |
| 02:19.59 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 02:20.32 | mackes | Hey Everyone |
| 02:23.51 | mackes | Hmmmm |
| 02:23.57 | mackes | No Chatter? |
| 02:24.08 | mackes | ~mackes |
| 02:24.09 | jbot | from memory, mackes is the man |
| 02:24.10 | denon | we're all talking about you in privmsg |
| 02:24.17 | mackes | Ahhh. |
| 02:24.21 | mackes | all |
| 02:24.26 | mackes | 273? |
| 02:24.31 | denon | yes, multicast privmsg |
| 02:24.32 | mackes | That is cool |
| 02:24.35 | mackes | Nice |
| 02:24.39 | mackes | ##asterisk |
| 02:24.56 | denon | #asterisk-lynch-mob |
| 02:25.10 | mackes | Hmmm |
| 02:25.24 | mackes | Created a channel and everything huh? |
| 02:25.44 | denon | been there ever since you came in |
| 02:26.10 | mackes | fast work. |
| 02:27.51 | trnzmeta | wohooo yah mofos, asterisk migration done :) |
| 02:28.04 | mackes | From what to what? |
| 02:28.10 | mackes | 1.2 to 1.4? |
| 02:28.14 | mackes | Server to Server? |
| 02:30.03 | [TK]D-Fender | I was a mofo once.... then realized I wanted my own instead of it coming bundled with the package ;) |
| 02:31.02 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 02:34.44 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
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| 02:36.38 | *** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net) |
| 02:38.18 | *** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net) |
| 02:47.57 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 02:48.13 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 02:54.50 | sevard | Does anyone remember the signate classical music on hold files? They had them in ulaw and mp3, royalty free classical music but now the website is down.. |
| 02:54.55 | sevard | does anyone have the MP3s? |
| 02:59.50 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 03:00.25 | watchy | i have some users who forward there polycoms to their cell phones, how do i stop them from getting calls when i ring a call group? |
| 03:00.37 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 03:03.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:05.57 | [TK]D-Fender | watchy, make sure they aren't part of one. |
| 03:06.43 | watchy | hrm ok but check this |
| 03:07.04 | watchy | say i have 5 idiots with phones who are in the office alot, and they need to ring when the IT guys are called |
| 03:07.14 | [TK]D-Fender | watchy, In this case it means either a) no ring-group membership ; b) 2nd reg for ring-group purposes ; or c) force your users to use dialplan based forwarding and not phone-based |
| 03:07.19 | watchy | so one can pick it up, well when one leaves the office he forwards it to his cell |
| 03:07.42 | watchy | which would you implement? |
| 03:07.43 | *** join/#asterisk killmel8tr (n=IceChat7@c-69-244-155-174.hsd1.mi.comcast.net) |
| 03:07.59 | [TK]D-Fender | watchy, "C" |
| 03:08.27 | watchy | so youd write a dialplan to check to see if that phone was forwarded |
| 03:08.30 | watchy | before ringing it? |
| 03:09.18 | watchy | im trying to learn as much as i can tk. so if im wrong just call me a idiot |
| 03:10.28 | [TK]D-Fender | watchy, Ok... you're an idiot :p You wouldn't have to worry about ringing it in a group because only DIRECTED calls on extens that WOULD care would have this check. |
| 03:12.07 | watchy | is there like a document about it i could read? |
| 03:18.54 | [TK]D-Fender | ~book |
| 03:18.55 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:19.32 | watchy | what should i look up, i owns that book |
| 03:19.32 | watchy | haha |
| 03:19.56 | [TK]D-Fender | watchy, this is all just dialplan logic with GotoIf's & astDb. Use some IQ here... |
| 03:20.12 | watchy | dont make me beat you to death tk |
| 03:20.18 | watchy | i can do it with help |
| 03:20.24 | watchy | beat you up that is |
| 03:20.45 | [TK]D-Fender | watchy, Oh yes.. you definitely need help, but I can't legally prescribe drugs for your "condition" ;) |
| 03:21.00 | watchy | so you just check to see if the ext is forwarded before dialing it right? |
| 03:21.15 | [TK]D-Fender | watchy, and I far too well armed & trained for that to be something you'd want to try ;) |
| 03:21.15 | watchy | i'm about to go pickup my book and start reading |
| 03:21.48 | [TK]D-Fender | watchy, Well go get reading, its only 3 really relevent dialplan apps. |
| 03:22.03 | watchy | one quick question |
| 03:22.33 | watchy | they pretty much would step #1 pickup fone, dial #888 or some # anyways then enter a # to forward to |
| 03:22.42 | watchy | and it would store it in * astdb |
| 03:22.47 | watchy | am i on the right track? |
| 03:22.56 | watchy | if so thats my last Q |
| 03:23.50 | [TK]D-Fender | watchy, pretty much. |
| 03:23.59 | watchy | thanks, thats all i needed. i'm off to ready homie |
| 03:24.01 | watchy | erad |
| 03:24.02 | watchy | read |
| 03:24.05 | [TK]D-Fender | watchy, give them extens to set/clear their forwarding, etc |
| 03:24.14 | watchy | yea |
| 03:24.24 | ManxPower | Oh Zaptel, how I shall miss thee! |
| 03:30.21 | mackes | All Snom Firmware |
| 03:30.24 | mackes | http://snom.provu.co.uk/sw/ |
| 03:41.04 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:42.02 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 03:57.56 | outtolunc | wonders if the tech type will be skinable .. meaning can i use Duh/ <G> |
| 03:58.30 | outtolunc | some might prefer Doh/ |
| 04:01.17 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 04:05.33 | outtolunc | gotta love mondays <G> |
| 04:05.46 | fujin | hehe |
| 04:05.50 | fujin | Gotta love tuesdays! |
| 04:08.44 | *** join/#asterisk wordzilla (n=me@122.109.125.252) |
| 04:12.16 | fujin | so, anyone |
| 04:12.28 | fujin | do I need files in /var/spool/asterisk/voicemail if I use odbcstorage AND configuration? |
| 04:12.35 | fujin | I mean, if I delete them, will everything still work? |
| 04:20.52 | *** join/#asterisk fiddur (i=fiddur@78.82.252.209) |
| 04:21.40 | craigk | i have a quick question, which may be silly - so pleaes be gentle :). I want to add support for a BRI ISDN line to my asterisk system, should I use mISDN channel or CAPI ? |
| 04:25.56 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 04:34.56 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 04:39.05 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 04:39.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:39.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:51.33 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 04:51.37 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:51.37 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:53.41 | *** join/#asterisk kasscie (n=ymonsalv@200.27.102.134) |
| 04:53.48 | kasscie | hi |
| 04:54.00 | kasscie | help me please |
| 04:54.48 | kasscie | i need configure iax2 for connect to asterisk |
| 04:55.21 | kasscie | my problema is one server asterisk is behind NAT and host dynamic |
| 04:56.48 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:03.09 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 05:03.55 | kasscie | mmmmmmmm |
| 05:06.32 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 05:07.56 | *** join/#asterisk beernuts (n=mattfox7@58.170.70.122) |
| 05:08.34 | beernuts | We have just upgraded to 1.4.20-rc1 to resolve a manage redirect issue and found that the following method to turn off music on hold no longer works, it just reverts to the default moh class which we use for our IVR. Basically we are trying to turn off the MOH for agents and even playing a silence sound file produces noise on the line that you can hear in a quiet environment. |
| 05:08.35 | beernuts | Does anyone have any other suggestions to completely stop playing a file for moh? |
| 05:11.02 | tengulre | kasscie: and then? |
| 05:14.08 | [TK]D-Fender | kasscie, setup a dyndns service on that server and use the hostname. |
| 05:17.05 | [TK]D-Fender | ok, checkout time. Later all |
| 05:21.09 | *** join/#asterisk webman (n=adamg@124.246.8.196.static.nexnet.net.au) |
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| 05:44.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:54.34 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 05:54.45 | watchy | would i need multiple servers with a 300 handset * install? |
| 05:55.28 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 06:03.24 | *** join/#asterisk CCFL_Man2 (i=b5da5a59@pool-70-105-211-134.scr.east.verizon.net) |
| 06:03.42 | CCFL_Man2 | anyone good with cisco voip dial peers? |
| 06:03.44 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 06:05.03 | CCFL_Man2 | anyone want to send a test sip call to me? |
| 06:10.28 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 06:19.21 | *** join/#asterisk olinux (i=olinux@ip68-6-186-151.sd.sd.cox.net) |
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| 06:59.09 | trnzmeta | anyone know of good softphone with ability to develop |
| 06:59.14 | trnzmeta | API and doco |
| 07:01.21 | fiddur | wengophone seemed ok... |
| 07:02.24 | kasscie | exist softphone o SIP UA in php? |
| 07:02.44 | webman | anyone use H323 with asterisk 1.4 ? I've noticed recently that "netstat -an" shows an increasing number of TCP ports in "LISTEN" status.... |
| 07:03.44 | webman | sorry, ports that are in LISTEN status are rtp ports... but I think they should be cleaned up, not continue to get more ports in LISTEN mode ? |
| 07:06.10 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:06.33 | kasscie | exist softphone o SIP UA in php? |
| 07:09.29 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 07:13.58 | *** join/#asterisk xenonex (n=xenonex@89.218.233.68) |
| 07:14.37 | kasscie | s |
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| 07:36.11 | r0land | hello all |
| 07:36.30 | r0land | could some1 help me out with a "503 service unavailable" error im having on my sip debugging screen! |
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| 07:38.08 | kasscie | s |
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| 07:41.32 | CCFL_Man2 | hah, infomercial for magic jack |
| 07:41.39 | CCFL_Man2 | so funny |
| 07:47.46 | CCFL_Man2 | audio quality sucks, the computer monitor interfears with the camra |
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| 07:55.25 | CCFL_Man2 | this majic jack infomercial is such a joke |
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| 08:10.17 | r0land | anyone could point me out to a good configuration for PSTN line interface on sipura 3102! |
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| 08:36.21 | *** join/#asterisk hoagie|ast (n=andi@numenor.segfault.net) |
| 08:38.03 | hoagie|ast | i've two isdn bri lines in te mode and i want to use the two ports in p2p mode and enable did. is it better to use a beronet card with misdn or a junghanns card with bristuff? |
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| 08:55.07 | mvanbaak | hoagie|ast: use the digium bri card |
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| 08:57.05 | hoagie|ast | you mean the b410p 4 port bri? |
| 08:57.16 | hoagie|ast | because they don't have a 2 port bri it seems |
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| 09:01.07 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 09:02.37 | Faustov | hi, i'm having a problem with exten => t,1,goto(#,1) at the end of my voicemenu - everything is being executed until this line, instead of looping the menu i get "Auto fallthrough, channel 'SIP/0012-c0029e90' status is 'UNKNOWN'" - why could this be? I just eliminated the timeout cause |
| 09:03.08 | yang | hoagie|ast: sweex PCI 128k cards are around 20 eur each |
| 09:04.02 | yang | hoagie|ast: They got HFC-4S chip on them, and work well with ISDN |
| 09:05.37 | hoagie|ast | and they work in ptp mode with te? |
| 09:05.45 | yang | hoagie|ast: haven't tried that |
| 09:06.15 | yang | hoagie|ast: they work in TE |
| 09:06.29 | hoagie|ast | ok thanks |
| 09:06.47 | yang | hoagie|ast: I don't know anything about ptp mode |
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| 09:07.54 | yang | hoagie|ast: if you go to the shop, simply ask for ISDN modem card, most of them have the HFC chip which is a good one |
| 09:08.34 | hoagie|ast | is this hfc chip that one by siemens? |
| 09:08.43 | florz | hoagie|ast: nope |
| 09:08.49 | yang | collognechip.de is the producer |
| 09:08.55 | florz | -l |
| 09:09.58 | Faustov | "Auto fallthrough occurs when there is nothing left for an extensions context to do, and a dial fails." - so why doesn't it execute the event for timeout specified by t,1,goto...? |
| 09:10.52 | creativx | t, timeout |
| 09:11.36 | Faustov | yeah, so what i'm trying to get is: when there's timeout, play the menu from the beginning again (loop) |
| 09:12.01 | Faustov | isn't exten => t,1,Goto(voicemenu,s,1) the right solution? |
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| 09:13.15 | hoagie|ast | thanks |
| 09:13.23 | hoagie|ast | i'll have a look at it |
| 09:16.08 | *** join/#asterisk BrokenNoze (n=Simon_Kr@host86-150-237-169.range86-150.btcentralplus.com) |
| 09:16.38 | BrokenNoze | Hi all, anyone know how to turn off the call waiting beep on a Polycom 650? keeps beeping in peoples ears half way through a call |
| 09:21.47 | *** join/#asterisk marl (n=marl@84.13.35.31) |
| 09:24.08 | marl | hi folks, anyone know if its posible to trace extensions.conf? i have upgraded from 1.4 to 1.6, and ive ound that my disa doesnt work anymore, was wanting to try and figgure were it was failing, basicly i dial in from my mobile, the CID authenticates me, and calls DISA, i get my dial tone on the mobile, but cant dial any numbers, if i connect directly to my internel context (the one that disa goes to) i can dial without any probs |
| 09:24.31 | Faustov | ok fixed, autofallthrough has to be set to no |
| 09:24.34 | Faustov | or t is being ignored |
| 09:25.27 | marl | autofallthrough is set to yes in extensions.conf |
| 09:25.27 | Strom_M | Faustov: that's not a fix; that's a kludge |
| 09:25.53 | marl | sorry thougth that was for me :) |
| 09:26.03 | Strom_M | marl: connect to the CLI, set the verbosity up, and watch |
| 09:27.07 | Faustov | Strom_M: well for my setup it's a fix, for asterisk it's just a setting :P |
| 09:27.31 | Strom_M | Faustov: you're ignoring the fact that a good dialplan should work properly with auto-fallthrough sey |
| 09:27.33 | Strom_M | er, set |
| 09:27.49 | Strom_M | falling off the end of an extension and then just waiting for a timeout to occur is a security risk |
| 09:28.03 | marl | was using the CLI, it just shows the DISA command being executed, as i get my dial tone, and hen nothing until the call times out :( |
| 09:29.06 | Faustov | Strom_M: could you explain? |
| 09:30.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:30.45 | Faustov | another question - is there a limit of simultaneous meetme participants? |
| 09:30.55 | Faustov | configured by zapata.conf |
| 09:35.14 | *** join/#asterisk bmg505 (n=leon@196-209-78-202-tbnb-esr-2.dynamic.isadsl.co.za) |
| 09:35.38 | marl | is there a way of getting docs for * apps like disa? |
| 09:36.50 | marl | (other than voip-info.org) |
| 09:38.04 | marl | as the 1.6 upgrade file mentions 5 args for disa, but the only docs i can find are on voip-info and they only mention 2 args |
| 09:39.50 | *** join/#asterisk wordzilla (n=me@122.109.125.252) |
| 09:40.53 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.202) |
| 09:40.59 | bartpbx | hello |
| 09:41.22 | bartpbx | We've just updated one of our servers to Version 1.2.28 |
| 09:41.46 | bartpbx | no the server has a load of more than 1 with only 50-60Calls |
| 09:43.01 | bartpbx | I could not identify any issue in the changelog that could cause this |
| 09:43.17 | *** join/#asterisk webman (n=adamg@200.179.233.220.exetel.com.au) |
| 09:43.18 | bartpbx | anyone has seen this? |
| 09:43.58 | webman | see what ? |
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| 09:44.23 | bartpbx | We've just updated one of our servers to Version 1.2.28 |
| 09:44.32 | bartpbx | now the server has a load of more than 1 with only 50-60Calls |
| 09:44.36 | [hC] | bartpbx: negative. asterisk is the process doing it? |
| 09:44.37 | bartpbx | that |
| 09:44.39 | webman | oh, sorry I don't use 1.2.x |
| 09:44.55 | bartpbx | hC: yes |
| 09:45.08 | bartpbx | the asterisk process takes all this |
| 09:45.15 | [hC] | what version were you at before? |
| 09:45.25 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:45.26 | bartpbx | 1.2.24 |
| 09:45.40 | bartpbx | top shows more than 100% from time to time |
| 09:45.41 | [hC] | very strange |
| 09:45.45 | bartpbx | currently 124% |
| 09:45.49 | bartpbx | yes |
| 09:45.52 | bartpbx | absolutly |
| 09:45.57 | webman | bartpbx: what mix of channels do you use ? |
| 09:46.07 | bartpbx | only sip + iax |
| 09:46.12 | bartpbx | more than 95% sip |
| 09:46.52 | RoyK | bartpbx: is that asterisk eating all that? |
| 09:46.57 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 09:46.58 | webman | if you restart asterisk is it easily reproducible ? |
| 09:47.12 | bartpbx | hm. i'll try tonight |
| 09:47.13 | RoyK | bartpbx: is zaptel involved? how many concurrent users+ |
| 09:47.14 | RoyK | ? |
| 09:47.26 | bartpbx | royk only sip + iax channels |
| 09:47.36 | bartpbx | currently ~ 50 Calls |
| 09:47.41 | bartpbx | on this maschine |
| 09:48.07 | webman | does anyone use H323 with asterisk 1.4 and lots of concurrent channels ? (over 100?) |
| 09:48.11 | bartpbx | this maschine handeld more than 100Calls with 1.2.24 and a load unter 1 |
| 09:48.47 | RoyK | bartpbx: have you upgraded recently? |
| 09:48.54 | bartpbx | yes |
| 09:48.58 | RoyK | :) |
| 09:48.59 | webman | for me, asterisk seems to have 'extra' TCP ports in LISTEN mode.... |
| 09:49.03 | RoyK | 1.4 or 1.6? |
| 09:49.06 | bartpbx | 1.2 |
| 09:49.23 | RoyK | ok. try downgrading, and stay there :P |
| 09:49.36 | bartpbx | haha |
| 09:49.40 | RoyK | with that load and 1.2 with no jitterbuffer, audio must be terrible |
| 09:50.13 | bartpbx | its a dual xeon maschine |
| 09:50.34 | RoyK | that comment on downgrading was meant seriously! |
| 09:51.09 | bartpbx | but we hat some issues with lockedup iax2 channels that forced us to the upgrade |
| 09:51.51 | RoyK | also - use my jitterbuffer patch from http://karlsbakk.net/asterisk/patches/1.2.x/rtp_jitterbuffer/ - it adds a jitterbuffer and stops asterisk from being utterly stupid and^W^W^W^Wrewriting the RTP timestamps |
| 09:52.21 | RoyK | asterisk 1.2 rewrites RTP timestamps while bridging, so dejittering later is impossible |
| 09:52.49 | RoyK | making audio quite terrible with SIP and jittery connections |
| 09:53.09 | bartpbx | how does 1.4 handle this? |
| 09:53.25 | [hC] | would that cause an issue im seeing with some people with jittery connections (comcast) wherein a call starts out okay but over time becomes noticeably lagged, in the form of up to a 2 second audio delay? |
| 09:53.28 | [hC] | Ive never understood that |
| 09:53.48 | [hC] | how a call can start out okay and after about 10 minutes be almost 2-3 seconds delayed and just keep getting worse. |
| 09:54.08 | bartpbx | for a production environment would you say 1.4 ist the right version? |
| 09:54.22 | bartpbx | lastime we hat a lot issues with 1.4 so we went back to 1.2 |
| 09:54.26 | [hC] | personally i am considering skipping 1.4 and going straight to 1.6 |
| 09:54.32 | [hC] | 1.4 has its own share of weird crap |
| 09:54.34 | RoyK | bartpbx: I'd guess so. I know others using 1.2, but with the jitterbuffer patch |
| 09:54.35 | [hC] | i still use 1.2 in production |
| 09:54.50 | RoyK | [hC]: and what wierd crap is in 1.6? |
| 09:54.54 | bartpbx | but 1.6 is not stable or? |
| 09:55.02 | [hC] | 1.6 is beta |
| 09:55.07 | RoyK | 1.4 is beta :P |
| 09:55.07 | [hC] | I havent touched 1.6 yet |
| 09:55.09 | bartpbx | is it stable? |
| 09:55.24 | RoyK | beta != stable |
| 09:55.27 | [hC] | 1.4 is tagged as stable, but imho its still got a lot of stuff in it that makes it difficult to just roll out |
| 09:55.36 | [hC] | for example rfc2833 dtmf |
| 09:55.39 | RoyK | [hC]: 1.4 is tagged as 'release', not stable :P |
| 09:55.42 | bartpbx | hehe |
| 09:55.54 | bartpbx | last time we used 1.4 it was far away from stable |
| 09:56.00 | [hC] | RoyK: touche. which is why i still have all my customers on 1.2 |
| 09:56.02 | [hC] | :) |
| 09:57.31 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:57.31 | RoyK | [hC]: still, 1.2 is quite useless without the jb patch |
| 09:57.46 | bartpbx | hC: how many calls does your 1.2 machines handle? |
| 09:57.49 | RoyK | [hC]: I paid some guy to write that some three years back or so |
| 09:57.54 | [hC] | RoyK: I dunno, I havent had a situation yet where a jitterbuffer would have saved the day. Maybe i just dont realize it? |
| 09:58.11 | [hC] | bartpbx: hmm, upwards of 50-60 |
| 09:58.12 | bartpbx | royk: does 1.4 /1.6 needs your patch as well? |
| 09:58.14 | RoyK | uses callweaver in production - t38gateway and all |
| 09:58.25 | RoyK | bartpbx: the jitterbuffer was merged before 1.4 |
| 09:58.30 | bartpbx | fine |
| 09:58.46 | RoyK | bartpbx: you might want to try callweaver. it forked from the 1.2 code |
| 09:58.46 | bartpbx | anyone using 1.6 in production? |
| 09:58.46 | [hC] | RoyK: is that the channel independent jitterbuffer, or just a fix for iax's jb? |
| 09:58.59 | *** join/#asterisk threewayone (n=hellowor@222.127.173.145) |
| 09:59.02 | threewayone | hi guys |
| 09:59.08 | RoyK | no - it's a generic implementation - not related to the iax jb |
| 09:59.10 | threewayone | im getting this error: /msg NickServ IDENTIFY |
| 09:59.23 | bartpbx | We've evaluated callweaver but we need to change a lot to get it running |
| 09:59.25 | threewayone | May 20 05:56:11 WARNING[32005]: chan_sip.c:9889 handle_response_invite: Forbidden - wrong password on authentication for INVITE |
| 09:59.34 | [hC] | t38gateway you say. |
| 09:59.37 | RoyK | asterisk supports using this generic jb on sip/rtp - callweaver supports this on all channel types |
| 09:59.40 | RoyK | [hC]: yes |
| 09:59.50 | [hC] | piques interest |
| 09:59.51 | *** join/#asterisk disposable (n=disposab@blackhole.sk) |
| 10:00.56 | threewayone | May 20 05:56:11 WARNING[32005]: chan_sip.c:9889 handle_response_invite: Forbidden - wrong password on authentication for INVITE --> im getting this error, my voip provider uses ip based authentication |
| 10:01.02 | *** join/#asterisk s0lid (n=s0lid@124.106.208.214) |
| 10:01.32 | bartpbx | anyone running 1.6 in a production env? |
| 10:03.41 | marl | bartpbx, was trying to run it, but hitting a few problems, so going back to 1.4 |
| 10:04.07 | marl | disa was causng problems, and there were a few other things that didnt seem to run corectly |
| 10:05.27 | threewayone | May 20 05:56:11 WARNING[32005]: chan_sip.c:9889 handle_response_invite: Forbidden - wrong password on authentication for INVITE --> im getting this error, my voip provider uses ip based authentication |
| 10:06.36 | RoyK | http://contexts.org/socimages/2008/05/16/icelandic-obamahillary-ad-obama-as-savage-cannibal/ |
| 10:07.23 | *** join/#asterisk wordzilla (n=me@122.109.125.252) |
| 10:12.06 | Falle | Has there been some change to how voicemail works? I upgraded to 1.4 and now when i call my extension that leads in into my nibox without password it just tells me that there is no new messages and then "goodbye" and hangs up even though i have old messages that i might want to handle. |
| 10:18.26 | bartpbx | marl: are you happy with 1.4 in prodction? |
| 10:18.39 | bartpbx | what channels are involved and how many calls? |
| 10:19.58 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 10:20.02 | r0land | hello all |
| 10:20.09 | r0land | anyone have experience with sipura 3102 ? |
| 10:22.17 | Faustov | which line in sip debug has the info what dtmf mode is in use when there's an incoming call? |
| 10:22.22 | Mw3 | hello. i just installed an asterisk system with a bri card. there is two S interface on the NT. one cable goes to the asterisk server and one to a bank pos terminal for credit cards. |
| 10:22.37 | Faustov | Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) <-- this one? |
| 10:23.04 | Mw3 | but either the pos terminal or the asterisk works. but not the two at the same time |
| 10:23.06 | _Raptor_ | r0land: the echo cancelation is crap |
| 10:23.21 | r0land | _Raptor_ i agree.. but i just wanna make it work till i get my digium card |
| 10:23.26 | r0land | _Raptor_ i cant even call out |
| 10:23.48 | r0land | Faustov ur asking me ? |
| 10:24.22 | Faithful | Mw3: I thought S interface had to be daisy chained? |
| 10:24.43 | _Raptor_ | r0land: oh i see, yes a digium card is the better choice |
| 10:24.45 | *** join/#asterisk kannan (n=kannan@123.201.60.110) |
| 10:25.15 | Faustov | r0land: anyone who knows |
| 10:25.18 | r0land | _Raptor_ ya i agree.. but meanwhile im beating myself up to get this sipura to work! whenever i try to call out from my softphone it gives me 503 service unavailable |
| 10:25.20 | _Raptor_ | r0land: du do not use this for calling out but only to make a normal analog phone work with asterisk |
| 10:26.20 | r0land | _Raptor_ well i have 4 pcs connected to lan which has the asterisk server as well as the Sipura on it.. so i got all the sftphones registered to asterisk.. as well as spa! so i want to b able to call eachother (i could do so..) and call outside through the pstn line |
| 10:26.49 | kannan | hello all, i m trying a quad span e-1 card with 2 pri lines in span 1 and 2. when i start asterisk i am continously getting warnings unable to disable echo cancellation on channel number -- , write to 137 failed unknown error 500 |
| 10:27.09 | r0land | _Raptor_ im having trouble calling outside through pstn, as well as recieving calls.. |
| 10:28.20 | Mw3 | Faithful: can you explain in more detail please? |
| 10:29.30 | JT | Mw3: asterisk cannot share S interfaces afaik |
| 10:29.43 | _Raptor_ | r0land: hmm i don't really know how to help you. i didn't touch the config for over a year because it worked :-) |
| 10:29.54 | r0land | _Raptor_ hehe lucky u.. |
| 10:30.09 | r0land | _Raptor_ how are u setup? digium.. or?! |
| 10:30.36 | kannan | i ave a digium TE410P |
| 10:31.23 | _Raptor_ | r0land: no arcor sip provider |
| 10:31.37 | r0land | _Raptor_ well could i have a peak to ur extensions.conf ? |
| 10:31.46 | r0land | and in sip.conf.. type= / |
| 10:32.07 | _Raptor_ | r0land: for the sipura? |
| 10:32.11 | r0land | _Raptor_ ya |
| 10:32.23 | threewayone | May 20 05:56:11 WARNING[32005]: chan_sip.c:9889 handle_response_invite: Forbidden - wrong password on authentication for INVITE --> im getting this error, my voip provider uses ip based authentication |
| 10:32.49 | _Raptor_ | r0land: sure, one sec |
| 10:32.56 | r0land | _Raptor_ thank u |
| 10:32.56 | Oy90 | threewayone: your provider requires password |
| 10:33.44 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 10:34.12 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:35.43 | _Raptor_ | r0land: http://pbot.rmdir.de/37298df0d1ca283fe0ee82b9f7b4729a extensions.conf |
| 10:35.48 | _Raptor_ | r0land: do you speak german? |
| 10:36.01 | r0land | _Raptor_ hate to disapoint u.. but no i dont |
| 10:36.48 | r0land | _Raptor_ so sipura_line is the context u got the extensions in sip.conf on ? |
| 10:36.56 | _Raptor_ | r0land: it says Macro(say,Please enter your phone number) |
| 10:37.07 | _Raptor_ | r0land: this collects the number via dtmf |
| 10:37.12 | r0land | _Raptor_ ah ok |
| 10:37.13 | _Raptor_ | r0land: yes |
| 10:37.34 | r0land | _Raptor_ well lemme give it a try maybe its my extensions.conf thts messed up |
| 10:37.46 | _Raptor_ | r0land: http://pbot.rmdir.de/0898a501e32c41dea3dfe4ed3f9639cb |
| 10:38.07 | _Raptor_ | r0land: you have to specify the sipura-line somewhere in the sipura config |
| 10:38.16 | r0land | yep of course |
| 10:38.18 | r0land | let me check it out |
| 10:38.28 | r0land | ill getb ack to u if it worked :) |
| 10:38.34 | *** join/#asterisk mort___ (n=mort@user-514f6626.l4.c3.dsl.pol.co.uk) |
| 10:38.57 | _Raptor_ | r0land: good luck :-) |
| 10:39.13 | r0land | thanks :) need it.. |
| 10:44.29 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
| 10:44.36 | r0land | _Raptor_ didnt work |
| 10:44.48 | r0land | _Raptor_ theres something in sipura.. its not detecting the line |
| 10:45.27 | _Raptor_ | r0land: mom, line means incoming! |
| 10:45.37 | r0land | i know |
| 10:45.39 | r0land | _Raptor_ i called in |
| 10:45.44 | _Raptor_ | oh |
| 10:45.48 | r0land | it kept on ringing without nothing happening on the asterisk CLI |
| 10:57.09 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:01.22 | threewayone | May 20 05:56:11 WARNING[32005]: chan_sip.c:9889 handle_response_invite: Forbidden - wrong password on authentication for INVITE --> im getting this error, my voip provider uses ip based authentication |
| 11:02.53 | Oy90 | threewayone: ask your provider |
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| 11:14.33 | *** join/#asterisk deltaray2 (n=Deltaray@68-76-215-54.cookmedical.com) |
| 11:15.29 | deltaray2 | I'm wondering, I've noticed that some companies are able to have their AA answer the phone before it even rings once. How are they doing this? Right now my it takes one ring before my asterisk server picks up. |
| 11:16.29 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 11:21.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:23.57 | florz | deltaray2: why should an AA "ring" before answering!? |
| 11:24.37 | deltaray2 | No I mean if I call my asterisk server from the outside, there is one ring before asterisk picks up. |
| 11:24.50 | deltaray2 | But some places I call, there is no ring. |
| 11:24.59 | florz | well, what do you mean by "ring"? |
| 11:25.12 | *** join/#asterisk Defraz (n=T0tal@65.39.96.74) |
| 11:26.41 | deltaray2 | florz, ??? Um, the ring sound that you hear repeatedly when you call someone. You know, Brrrrring! Brrrrrring! What else could I mean? |
| 11:27.24 | florz | deltaray2: well you could mean the sound that a telephone make when trying to make you aware that someone is calling, for example |
| 11:27.40 | deltaray2 | Ok, looking at Wikipedia, maybe you want me to distinguish that I'm talking about the ringback signal as opposed to a ringtone I'd hear when someone calls me. |
| 11:28.27 | deltaray2 | Anyways.... |
| 11:28.48 | florz | maybe - and that's a pretty important distinction as one doesn't have much to do with the other |
| 11:28.49 | deltaray2 | How are they having their phone system answering before you hear a ringback tone? |
| 11:29.03 | florz | except that you usually get a ringback while a phone is ringing |
| 11:29.32 | deltaray2 | Ok, anyways... I think I was clear as to what I was talking about from my question. Do you know the answer to my question? |
| 11:29.35 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:29.54 | florz | Well, why shouldn't they? If they know a call is comming in, why should they wait until you have heard some ringback or something? How should they even know what you hear? |
| 11:30.05 | florz | -m |
| 11:31.13 | deltaray2 | Do you enjoy derailing people's questions? |
| 11:31.28 | florz | nope, not at all |
| 11:31.44 | deltaray2 | Then why do it? |
| 11:31.48 | florz | I don't |
| 11:31.58 | florz | at least not that I was aware |
| 11:32.42 | deltaray2 | You don't have an answer for my question except for some sarcastic remark, you pick apart the semantics of what I'm asking as if it wasn't obvious that I was talking about calling someone and not receiving a call, etc... |
| 11:33.13 | florz | I guess you simply don't understand what I am saying |
| 11:33.32 | florz | you do correlate things that don't have much to do with one another |
| 11:33.49 | florz | and all I am sayin, is, basically, that you shouldn't |
| 11:34.00 | deltaray2 | If you would read back what you've told me, it essentially amounts to nothing helpful. |
| 11:34.21 | deltaray2 | Ok, so you're saying that I shouldn't be hearing a ringback tone when I call my asterisk server? |
| 11:34.23 | florz | Well, that depends on what problem you are trying to solve, I guess |
| 11:34.37 | florz | Erm, no!? |
| 11:35.42 | deltaray2 | Ok, let's forget everything that's been said above and start over. I'll ask a question. |
| 11:36.07 | deltaray2 | How can I make it so that my asterisk server picks up before the caller hears any ringback tone? |
| 11:36.37 | Uatec | Answer() |
| 11:36.38 | Uatec | ? |
| 11:36.39 | florz | you can't, pretty much |
| 11:37.25 | deltaray2 | Ok, but some other people are doing this with their phone systems. When I call them, there is no ringback tone before their system picks up. They may not be using Asterisk. Is this a limitation of Asterisk? |
| 11:38.15 | florz | All you can do is answer as quick as possible, basically, and then hope that noone started playing a ringback tone before you picking up is signalled to them. |
| 11:39.19 | florz | Plus, depending on the operators and kind of technology involved, you may be able to send any alerting messages (or equivalent) only shortly before picking up |
| 11:39.21 | *** join/#asterisk blinky42 (n=blinky42@67.106.175.146.ptr.us.xo.net) |
| 11:39.53 | deltaray2 | Ok, so basically, you don't know how this is done. |
| 11:40.31 | florz | well, obviously not for some specific case |
| 11:40.44 | florz | this isn't somehow a feature of telephone networks, usually |
| 11:41.18 | deltaray2 | Ok, but there must be some way that people are doing it. Does it maybe have something to do with being connected more directly to the PSTN, like through a T1 instead of a POTS line? |
| 11:41.28 | florz | it's just that, depending on the devices and protocols used, you may be able to time signalling such that there isn't any/much ringback to be heard |
| 11:41.38 | deltaray2 | I can give you an example of a company that is doing this if you want and their phone number. |
| 11:42.14 | deltaray2 | Hmmm. Ok. Well if its not standard, I don't think I'd want to do it, but I was mostly just curious as to how its done. |
| 11:43.00 | florz | Well, POTS is a kind of protocol(s) that usually won't have any way to influence the timing of call progress signalling, except for pickup and hangup |
| 11:45.27 | florz | Well, in principle, what happens there, is that a message is sent in the direction of the callee, and then is routed through some network of telco equipment, forwarded hop by hop, until it reaches some entity that can decide whether the dialed number does exist |
| 11:45.58 | florz | and you obviously don't want to play ringback until you know whether a number does exist |
| 11:46.50 | florz | so the initiating end waits for a reply that says "yep, found that endpoint, telling it that you have a call for it now" |
| 11:47.08 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 11:48.33 | florz | but if the entity sending that message and the actual endpoint somehow work closely together/are the same device/..., they obviously can send back-to-back messages "yep, found that endpoint" and "endpoint has picked up", thus resulting in 0 length ringback |
| 11:48.42 | florz | at least usually ... |
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| 11:49.29 | deltaray2 | Ok, well that's possible since at the company that I know that does this is an ISP/VoIP provider. |
| 11:51.03 | florz | (however, you usually can't, for example, create a few minutes of ringback-free "ringing", as the calling end usually will have a relatively short timeout that it waits for an answer telling it whether the number does exist) |
| 11:52.13 | *** join/#asterisk mort_gib (n=mjensen@dsl-p5-219.gibconnect.com) |
| 11:53.45 | kannan | hello, on my TE410p , in san one i have connected the pri span. before that i did the patloop test , which ran fine. the LED on the cad is green. Bt when i go the * cli, i have a large number of alram detected, alarm cleared messages running all along. I am not ale to call out thru the pri. SIP calls are fine |
| 11:58.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:58.36 | kannan | the company here says they made cal, y'day from a pbx on the pri. Am i right in thinking it a service telco provider end issue? |
| 12:03.41 | deltaray2 | florz, Just curious, where are you from, North America? |
| 12:04.08 | florz | deltaray2: .de |
| 12:04.16 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 12:05.56 | kannan | i am sure my conf files are correct as i have other aserisk servers working with the same telco |
| 12:06.02 | deltaray2 | florz, ok, then it would stand to say that you're used to a different type of phone system. |
| 12:07.13 | deltaray2 | Maybe I should have waited to ask my question later in the day when more people from NA are on the channel. |
| 12:07.58 | JT | deltaray2: maybe you should stop talking down to everyone |
| 12:08.05 | florz | deltaray2: Well, that depends on the level at which you look at it. But your phone system over there really isn't that much different in general ;-) |
| 12:08.25 | JT | most phone systems are pretty similar around the world |
| 12:08.39 | JT | it'd quite arrogant to assume they're that different with no evidence of such |
| 12:10.35 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 12:11.50 | JT | but in reality it's quite easy |
| 12:11.52 | JT | get a PRI |
| 12:12.00 | JT | and answer calls immediately |
| 12:12.03 | deltaray2 | JT, and it would be arrogant on your part to assume I don't have any evidence. There are subtle differences and the question I asked could definitely depend on those subtle differences. |
| 12:12.21 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 12:12.22 | JT | most of the time callers won't hear ringing |
| 12:12.24 | deltaray2 | JT, Thank you, that answers my question. So this is something to do with having PRI/T1? |
| 12:12.29 | JT | but there's no guarantee |
| 12:12.37 | deltaray2 | Ok |
| 12:12.40 | JT | sure, you can't answer POTS fast enough |
| 12:12.46 | sysadmin-lb22 | hi asterisk is dying..with illegal instruction..Makefile is for the correct PROC version i.386 |
| 12:12.50 | JT | it has to ring at least once |
| 12:13.43 | r0land | hello all |
| 12:13.49 | r0land | could some1 help out with sipura 3102? |
| 12:14.20 | JT | you need a form of digital signalling that gives you call setup messages |
| 12:14.26 | JT | which is a PRI for most people |
| 12:14.26 | florz | JT: why not? you could probably use a single (half a) period of the ringing sine for detection?! But no clue whether any CO would recognize that before the ring is over ... ;-) |
| 12:14.54 | JT | VoIP is also possible, but lack of ringing would be less reliable there as it'd depend on your provider |
| 12:15.20 | JT | heh |
| 12:16.55 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:31.47 | MaartenB | hi guys, I have a lot of problems with transfering calls to another phone, the calling party gets dropped, even when canreinvite=no, any suggestions how to debug this? |
| 12:32.13 | kannan | hello, i am trying to build net for zttool. i get a error saying no lslang when i do make |
| 12:32.26 | kannan | newt, not net |
| 12:32.31 | *** join/#asterisk ManxPower (n=manxpowe@12.sub-70-220-153.myvzw.com) |
| 12:32.47 | Faustov | could someone please have a look at http://pastebin.com/me79a1c7 <-- this is the cli log from 2 calls, one successful into app-meetme and the other one fails at one point - as if # was being interpreted wrongly - why would this happen? |
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| 12:39.36 | dandre | Hello, |
| 12:40.22 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 12:40.25 | dandre | Is there any free development tool to build a tapi driver? I intend to write some tapi driver for asterisk |
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| 12:43.37 | kannan | ok i got newt running and zttol is showing ok |
| 12:44.17 | kannan | sometimes tho it changes to yellow and immediately back to ok. the CLI shows alarm detedcted, and he alarm cleared continually. notabe to make calls thru pri on the TE410P |
| 12:44.29 | kannan | no able |
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| 12:48.30 | kannan | can someone help with this please? |
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| 13:00.06 | ManxPower | kannan: put the contents of /etc/zaptel.conf on pastebin.ca |
| 13:00.42 | ManxPower | kannan: odd that you are not getting HDLC Abort errors with those Yellow alarms |
| 13:04.16 | kannan | ManxPower , sory was on the other screen , will do now |
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| 13:06.57 | Faustov | DYNAMIC_FEATURES=>automon#blindxfer <---- is this syntax correct? |
| 13:07.27 | Faustov | in [globals] in extensions.conf, that is |
| 13:09.03 | kannan | ManxPower http://pastebin.ca/1023760 |
| 13:09.46 | *** join/#asterisk mbranca (n=matteo@mi-gw1.voismart.net) |
| 13:09.55 | ManxPower | change this line "span=2,1,0,ccs,hdb3" to this "span=2,0,0,ccs,hdb3" |
| 13:10.09 | ManxPower | I assume span=1 is the telco or are both spans going to the telco. |
| 13:11.09 | ManxPower | kannan: we are volunteers here, so try not to be AFK for too long while we work on your problem. |
| 13:11.22 | cpm | chuckles |
| 13:11.36 | cpm | ManxPower, just get it done! what's wrong with you? |
| 13:11.41 | ManxPower | Faustov: in [globals] VARIABLE=VALUE not VARIABLE=>VALUE |
| 13:11.56 | ManxPower | Faustov: I don't think it matters, but it might. |
| 13:12.26 | Faustov | ManxPower: what about 2 values? |
| 13:12.44 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:13.04 | Faustov | ManxPower: i need automon and blindxfer, because with default setup the blindxfer app takes over the # key for transfer and i need it to finish recordings |
| 13:13.28 | *** join/#asterisk deltaray2 (n=Deltaray@68-76-215-54.cookmedical.com) |
| 13:13.54 | *** part/#asterisk deltaray2 (n=Deltaray@68-76-215-54.cookmedical.com) |
| 13:14.03 | ManxPower | You don't set a variable to more than one value. "automon#blindxfer" is ONE value. The applications break that out into different values. |
| 13:14.42 | Faustov | still doesn't help |
| 13:14.48 | Faustov | i don't get it |
| 13:15.26 | Faustov | ManxPower: how would you disable blindxfer? |
| 13:15.28 | ManxPower | Faustov: did you remember to use the w/W options to Dial? |
| 13:16.00 | Faustov | ManxPower: no no, automon works, i'm trying to prevent blindxfer from taking over the # key |
| 13:16.07 | ManxPower | Faustov: Are your phones really bad? most people use the built in transfer feature of their phone, not the DTMF transfer hack. |
| 13:16.32 | kannan | ManxPower : any ideas? This zaptel.conf works with this telco, (at other sites) |
| 13:16.35 | Faustov | ManxPower: no, i get this from all phones, mobiles or any |
| 13:16.45 | *** join/#asterisk thepacmanfan (n=thepacma@12-218-140-147.client.mchsi.com) |
| 13:16.47 | ManxPower | Faustov: remove the #blindxfer and don't use t/T on the Dial plan |
| 13:16.58 | kannan | both spans are going to the telco |
| 13:17.09 | ManxPower | kannan: yes. I gave you the change to make. |
| 13:17.21 | ManxPower | you'll have to run ztcfg -vvv and stop and start asterisk. |
| 13:17.33 | kannan | ManxPower thanx : will check this screen regularly , my apology |
| 13:18.02 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 13:19.20 | ManxPower | kannan: chances are you have a sync timing source problem. What channel numbers are going into yellow alarm? |
| 13:19.58 | ManxPower | Faustov: you really should read The Good Book again. |
| 13:20.00 | ManxPower | ~book |
| 13:20.01 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:20.55 | Faustov | ManxPower: i dont have t/T anywhere in my dialplan, i don't have blindxfer set anywhere but it still takes precedence when i need to save the recording |
| 13:21.13 | ManxPower | Faustov: then users should not be able to transfer. |
| 13:21.29 | ManxPower | Faustov: so you have a dialplan problem. |
| 13:21.41 | kannan | ManxPower : I am geting the same problem only |
| 13:21.53 | ManxPower | kannan: I do not understand |
| 13:22.09 | kannan | the span shos yellow alrm and hen geen imediately |
| 13:22.47 | ManxPower | kannan: BOTH spans? Only one span? Individual channels don't go into yellow alarm? |
| 13:22.54 | ManxPower | put a copy of the errors on pastebin.ca |
| 13:23.01 | kannan | from the cli? |
| 13:23.08 | ManxPower | correct |
| 13:23.12 | kannan | ok will do |
| 13:27.00 | kannan | ManxPower : http://pastebin.ca/1023775 - sorry its a copy paste from putty |
| 13:28.01 | ManxPower | kannan: it is looking more and more like a telco or cable issue. |
| 13:28.25 | kannan | i said as much. i had exactly the same prob oce and it was a cable prob in the mugcs |
| 13:28.42 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-a0db9861aa297df5) |
| 13:29.02 | kannan | but the company is not agrred with me, as y'day they made a call frm a tester given by the telco |
| 13:29.23 | kannan | the telco now asks , tell me what the exact problem is and we will see to it |
| 13:29.42 | kannan | the patlooptest runs fine on a loop back plug |
| 13:29.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:29.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:29.56 | ManxPower | kannan: to telco "I am getting a yellow alarm about every X mins" |
| 13:30.03 | kannan | and the zaptel.conf runs with telco on another asterisk box |
| 13:30.06 | ManxPower | kannan: you would want to try zttest too. |
| 13:30.13 | kannan | hmm x seconds is it |
| 13:30.23 | kannan | ManxPower , ok |
| 13:32.09 | kannan | ManxPower , ztest is mostly 100% but occasionally 99.987793% |
| 13:32.14 | kannan | zttest |
| 13:32.21 | ManxPower | kannan: those numbers should be OK |
| 13:32.24 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:33.04 | kannan | in fact after 181 passes that was the worst |
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| 13:33.46 | kannan | can i tell them to ask the telco to do test calls again from the site? |
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| 13:34.49 | kannan | ManxPower : I really appreciate the help you have given to me |
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| 13:49.23 | Faustov | could someone please have a look at http://pastebin.com/me79a1c7 <-- this is the cli log from 2 calls, one successful into app-meetme and the other one fails at one point - as if # was being interpreted wrongly - why would this happen? |
| 13:55.01 | b11d` | thats weird |
| 13:55.24 | b11d` | i see the local call and the cell call originate in different contexts.. |
| 13:56.17 | *** join/#asterisk xenonex (n=xenonex@92.47.1.182) |
| 14:00.45 | [TK]D-Fender | Faustov: considered answering that call before hitting MeetMe? |
| 14:01.19 | Faustov | [TK]D-Fender: why would that be any good? |
| 14:02.09 | [TK]D-Fender | Faustov: Just try it |
| 14:03.21 | JT | Faustov: from mobile phone? how does it get there? |
| 14:03.47 | Faustov | [TK]D-Fender: i'm not sure what you suggest here, should i transfer that call to the extension meetme has been assigned? |
| 14:04.29 | [TK]D-Fender | Faustov: This is not Raw-Cat Science. I said ANSWER the call before dumping them into MeetMe. |
| 14:06.09 | Faustov | JT: at the moment it's with the register => user:pass@provider/meetme-extension and then in extensions.conf i did exten => meetme-exten,1,MeetMe(meetme-exten,FMixq) |
| 14:06.27 | Faustov | [TK]D-Fender: oh i see, i'll try |
| 14:06.35 | JT | sip from a voip provider. |
| 14:07.50 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 14:07.54 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:09.12 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:11.15 | Faustov | [TK]D-Fender: same thing |
| 14:11.19 | Faustov | -- Executing [s@macro-conf:1] Answer("SIP/995229-007fc910", "") in new stack |
| 14:11.19 | Faustov | <PROTECTED> |
| 14:11.20 | *** join/#asterisk rati (n=rati@122.169.108.195) |
| 14:11.25 | Faustov | with answering |
| 14:11.36 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:11.37 | rati | Hey how can I check that my ISP has blocked VOIP services? |
| 14:13.18 | redback | call them |
| 14:13.28 | redback | (not via voip) |
| 14:13.32 | Faustov | :) |
| 14:13.35 | JT | Faustov: try hitting # on a call that doesn't go into meetme? |
| 14:13.44 | Faustov | JT: just did, same thing |
| 14:13.51 | JT | what happens? |
| 14:14.03 | Faustov | JT: i get "transfer" and "free line" signal |
| 14:15.46 | JT | ? |
| 14:16.02 | drehlecom | rati: http://www.cs.columbia.edu/sip/servers.html use one of them to test it |
| 14:16.35 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 14:16.45 | Faustov | JT: well # is used by default for blindxfer, but for some reason, even without enabling this feature in global, it takes precedence over the # used to end name recording in meetme |
| 14:17.18 | kannan | hello, the telco had a prob in his end, the TE410P is now working fine, thanks to all |
| 14:17.32 | kannan | esp ManxPower |
| 14:18.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:19.29 | thepacmanfan | anyone familiar with this error? channel.c:1865 ast_waitfordigit_full: Unexpected control subclass '2' |
| 14:20.16 | thepacmanfan | it occurs when putting an incoming analog call into a queue |
| 14:21.31 | drehlecom | rati: like http://interop.pingtel.com/ where you can use the test accounts |
| 14:21.54 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-152-139.dsl.teksavvy.com) |
| 14:23.46 | *** join/#asterisk mackes-Office (n=root@74.10.229.49) |
| 14:25.16 | Faustov | JT, [TK]D-Fender: i can get around this problem by disabling the names recording (q option for meetme) |
| 14:26.44 | Faustov | however there's another problem. I got dtmfmode=auto in global sip.conf, but people dialing in that meetme from abroad get invalid pin even tho they enter the valid pin |
| 14:27.12 | Faustov | what else could cause that, if not dtmf? |
| 14:28.44 | [TK]D-Fender | Faustov: if inband it could be distorted. |
| 14:29.33 | Faustov | [TK]D-Fender: is there any way i can find out? i tried comparing a successfull call and a failed one line by line with sip debug on |
| 14:29.51 | Faustov | but i couldn't see any difference, just one time pin accepted, the other time invalid |
| 14:30.10 | [TK]D-Fender | Faustov: make a small IVR to test DTMF reception. You shouldn't sue "auto" unless you have to anyways. You should find out what their mode is supposed to be and set accordingly. |
| 14:31.38 | *** join/#asterisk mort_gib (n=mjensen@8.Red-81-35-165.dynamicIP.rima-tde.net) |
| 14:33.02 | Faustov | [TK]D-Fender: the office in london claims to have "normal british telecoms phones", and their provider claims they don't change the dtmf |
| 14:33.17 | Faustov | [TK]D-Fender: does that mean "inband"? |
| 14:33.26 | [TK]D-Fender | Faustov: means NOTHING. |
| 14:33.44 | Faustov | heh true |
| 14:34.17 | [TK]D-Fender | Faustov: I get the uncontrollable urge to punch people how answer me like that :) |
| 14:34.20 | [TK]D-Fender | who* |
| 14:34.21 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:34.27 | disposable | is it Voicemail() or VoiceMail()? |
| 14:34.43 | [TK]D-Fender | disposable: Eiter. Apps aren't case-senesitive |
| 14:34.47 | [TK]D-Fender | either* |
| 14:35.01 | JT | Faustov: umm, you get calls via sip right? what codec? |
| 14:35.03 | [TK]D-Fender | Geez I'm out of typing sync today... |
| 14:35.21 | disposable | [TK]D-Fender: thanks. i've just realised i use both and both work |
| 14:35.23 | [TK]D-Fender | Faustov: Set for rfc2833, if it works, keep it. |
| 14:35.49 | Faustov | JT: yes, all 100% sip |
| 14:36.00 | JT | Faustov: codec? |
| 14:36.01 | Faustov | JT: ulaw |
| 14:36.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:36.18 | JT | Faustov: even from abroad, ulaw? |
| 14:36.34 | Faustov | JT: yeah, looks like it in the sip debig |
| 14:36.36 | Faustov | debug* |
| 14:36.39 | JT | Faustov: what country? |
| 14:37.24 | Faustov | uk -> poland |
| 14:37.37 | JT | ulaw, interesting choice |
| 14:37.46 | Faustov | why? |
| 14:37.50 | disposable | Faustov: definitely alaw |
| 14:38.01 | disposable | Faustov: nobody in europe uses ulaw |
| 14:38.06 | JT | alaw is the standard in almost every country outside usa and japan |
| 14:39.04 | Faustov | JT: ok, let me try |
| 14:39.15 | JT | this won't fix the dtmf issue though |
| 14:39.19 | JT | if that's even what it is |
| 14:40.57 | *** part/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 14:41.10 | Faustov | JT: just in case you wonder, it doesn't fix the "#" issue :> |
| 14:41.27 | JT | i knew it wouldn't, only a slight companding difference :) |
| 14:42.02 | *** join/#asterisk gardo (n=gardo@58.69.91.105) |
| 14:42.14 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 14:42.14 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:42.23 | [TK]D-Fender | Faustov: jitter/pl and general poor audio can all warp inband DTMF. Did you try rfc2833? |
| 14:43.05 | *** join/#asterisk daniev (n=psycke@201.245.223.90) |
| 14:43.51 | *** join/#asterisk juice_d (n=juice@home.tootight.net) |
| 14:44.05 | Faustov | [TK]D-Fender: i originally had rfc2833, but since the uk guys couldn't connect, i decided tochange this to auto |
| 14:44.23 | Faustov | [TK]D-Fender: back to rfc tho, because it didn't solve anything |
| 14:44.39 | daniev | hello guys |
| 14:44.52 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:45.13 | [TK]D-Fender | Faustov: So its on rfc2833 now and you still have a problem? |
| 14:45.55 | Faustov | [TK]D-Fender: yes, unfortunately |
| 14:46.07 | [TK]D-Fender | Faustov: Ok, thats the end of my advise then. |
| 14:46.23 | Faustov | [TK]D-Fender: could you maybe have a look at my extensions.conf and/or sip.conf? |
| 14:46.48 | daniev | my telephony provider(Telmex) is offering me two E1 options: rdsi pri net 5 and rdsi pri qsig. I've been searching info about differences between them with no success |
| 14:47.00 | *** join/#asterisk mort___ (n=mort@user-514f6626.l4.c3.dsl.pol.co.uk) |
| 14:47.09 | JT | daniev: country? |
| 14:47.18 | daniev | can anybody give me a short explanation to make a decision ? |
| 14:47.24 | daniev | JT: colombia |
| 14:47.40 | JT | both options, i don't know if support in asterisk is any good |
| 14:47.57 | JT | euroisdn, ni2, or dms would be much preferrable :) |
| 14:48.05 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:48.23 | daniev | JT: i know that one of those E1 types has been used with trixbox appliances |
| 14:48.47 | [TK]D-Fender | Faustov: No use. You say you get some DTMF but not all. That isn't a dialplan issue |
| 14:49.08 | [TK]D-Fender | Faustov: I suggest making an IVR for a full DTMF test just to see how bad it is. |
| 14:49.39 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:50.00 | webman | is anyone using asterisk 1.4.x to do around 100 concurrent H323 channels (just call in -> call out, not features on the box) with/without problems as mine seem to crash every day or two ? |
| 14:50.26 | [TK]D-Fender | webman: H.3232 + Crash = nearly synonymous. |
| 14:50.49 | daniev | JT: because the other option is R2 Q421. worst i guess. or am i wrong ? |
| 14:50.51 | JT | + Asterisk you mean |
| 14:51.08 | JT | daniev: MFC R2? run! |
| 14:51.18 | webman | so I should not expect to be able to use H323 in asterisk in a stable manner ? |
| 14:51.36 | JT | asterisk has appauling H.323 support |
| 14:51.42 | Faustov | [TK]D-Fender: just did, asked them to dial in and they had to pick "0" to connect to me, and they did |
| 14:51.50 | daniev | JT: really? that's the best ? |
| 14:52.05 | Faustov | [TK]D-Fender: so it's not dtmf, not codec, not dialplan - what else? bug in meetme? |
| 14:52.06 | daniev | talking about asterisk of course |
| 14:52.08 | JT | nothing that is an exact match for what asterisk does really |
| 14:52.20 | [TK]D-Fender | Faustov: of yeah.... 1 digit single extension with no repeat testing for QC. Yeah, thats thorough.... |
| 14:52.44 | [TK]D-Fender | </sarcasm> |
| 14:52.49 | Faustov | i got it :> |
| 14:52.56 | Faustov | thought 1 is enough |
| 14:53.04 | Faustov | so you say 1 could just be lucky :> |
| 14:53.32 | webman | JT: if enough info is provided, are there people who actively maintain (or at least bugfix) it ? ie, through the bugtracker. ... |
| 14:54.01 | JT | webman: i don't think there's even a native h.323 channel driver that's mantained |
| 14:54.09 | JT | there 3 asterisk h.323 channel drivers |
| 14:55.10 | webman | I'm using the one from the asterisk source tree... I assumed that would be the best maintained (at least, when I installed it, it had recent updates within the last 2 years).... |
| 14:55.32 | JT | try the others? |
| 14:55.59 | JT | or use something else to convert calls to sip first |
| 14:56.02 | [TK]D-Fender | webman: I know.... slap FreeSWITCH in front :) |
| 14:56.08 | Maliuta | Having defined things like mailbox in the sip channel definition can I access those things when the peer dials in? I know that callerid is passed, but I want to use the mailbox definition in the dialplan |
| 14:56.13 | JT | wrong answer, Yate :P |
| 14:56.15 | *** join/#asterisk s0lid (n=s0lid@58.69.91.105) |
| 14:56.26 | JT | freeswitch i don't think is quite there yet with H.323 |
| 14:56.52 | [TK]D-Fender | JT: ok, fine, sure, whatever :) |
| 14:57.29 | webman | well, all I am doing is receive the call, check the destination number in DB, and then pass the call to a quintum with H.323 ....is there some other tool which could do this? |
| 14:57.29 | JT | it seems to be a simple enough affair getting Yate going as a H.323 to SIP converter |
| 14:57.45 | JT | GnuGK |
| 14:59.56 | webman | yate is considered more reliable than asterisk at H.323 ? |
| 15:00.06 | JT | yes |
| 15:02.32 | fskrotzki | ~book |
| 15:02.33 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:03.32 | *** join/#asterisk mirrorcolor (n=iunixan@196.218.222.116) |
| 15:08.49 | webman | thanks for the info... I'll look into YATE and see how I go.... |
| 15:11.12 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 15:16.37 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 15:16.47 | *** join/#asterisk Wic3edOne (n=Wic3done@cpe-72-184-246-19.tampabay.res.rr.com) |
| 15:18.46 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:18.53 | *** join/#asterisk a-s (n=user@89.38.174.194) |
| 15:19.01 | a-s | hello |
| 15:19.20 | a-s | can somebody help me to install cdr ( http://areski.net/asterisk-stat-v2/about.php ), please? |
| 15:19.40 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 15:19.42 | b11d` | umm, |
| 15:19.46 | b11d` | are you sure you want to do that a-s? |
| 15:19.53 | b11d` | last release of it was in 2005 eh |
| 15:19.57 | b11d` | a touch outdated, no? |
| 15:20.34 | a-s | b11d`: you know a better and free analyser? |
| 15:21.15 | a-s | maybe it is stable, and it didn't need more updates |
| 15:21.17 | b11d` | i wrote my own custom CDR parser in perl.. |
| 15:21.25 | b11d` | but its hard coded to my environment |
| 15:21.50 | Shotygun | Anyone here knows about issue related to "ringinuse" ? For some reason even if ringinuse=no (or yes, for the sake of the discussion), the queue still rings at a busy SIP device |
| 15:21.57 | b11d` | i think asterisk has slighly changed CDR formats since 2005.. |
| 15:21.59 | b11d` | i could be wrong.. |
| 15:22.17 | putnopvut | Shotygun: do you have a call-limit set for the SIP device? |
| 15:22.34 | a-s | b11d`: I think you are wrong |
| 15:22.39 | a-s | look here |
| 15:22.40 | a-s | http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI |
| 15:22.51 | a-s | you can make the graphs work with PHP5 by downloading the latest version of jpgraph and dropping the source files in the jpgraph_lib directory. alternatively, downgrade to PHP4... |
| 15:23.15 | a-s | message from 2007 |
| 15:23.19 | b11d` | ok then im wrong.. go try it |
| 15:23.32 | b11d` | what are you even waiting for? |
| 15:24.01 | a-s | I do not know how to install a php page on apache :( |
| 15:24.01 | Shotygun | putnopvut: Yes, of 2. Although I rather have it "6" for personal reasons. the type is 'friend'. |
| 15:24.13 | b11d` | a-s then read the PHP and Apache manuals |
| 15:24.21 | Faustov | [TK]D-Fender: can we assume entering the same set of digits in the IVR as the pin number is - is a valid test of dtmf? |
| 15:24.21 | Shotygun | The thing is that I need them to receieve incoming calls when they are within a call, but not from the queue. |
| 15:24.41 | a-s | oh, I believed that that was easy, and somebody can tell me directly |
| 15:24.44 | [TK]D-Fender | Faustov: "no comment" |
| 15:24.44 | Maliuta | a-s: this is not the place to get apache/php help |
| 15:25.18 | putnopvut | Shotygun: that's why the ringinuse option is there. You can set a really high call-limit in sip.conf, but set ringinuse=no in queues.conf |
| 15:25.36 | putnopvut | Shotygun: that way, they can only receive one call at a time from the queue, but they can still receive other calls from elsewhere. |
| 15:25.44 | Maliuta | a-s: and I would suggest that if you can't setup and admin an apache server that * is way out of your league |
| 15:25.48 | Shotygun | putnopvut: That's what I did, it still rings at the member who is already talking with another call that came out of the queue |
| 15:26.12 | Shotygun | To where ast_log() goes? |
| 15:26.12 | Faustov | [TK]D-Fender: he also tried 1234, 3333, 8778 and he got thru all the menus (the uk guy) |
| 15:26.20 | putnopvut | Shotygun: did you set the call-limit within the peer in sip.conf? |
| 15:26.38 | Faustov | [TK]D-Fender: i think that's safe enough to say it's not dtmf? |
| 15:26.53 | Shotygun | putnopvut: Yes, should I remove it? |
| 15:26.54 | [TK]D-Fender | Faustov: You keep on thinking... |
| 15:27.10 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 15:27.14 | ccvp | heh, just realized this in the book: |
| 15:27.15 | putnopvut | Shotygun: no, you shouldn't...I don't know why that wouldn't be working...unless you're using realtime sip peers. |
| 15:27.16 | ccvp | As you explore the community, |
| 15:27.18 | ccvp | though, be aware that there are people with many different skill sets and attitudes here. |
| 15:27.20 | ccvp | Some of these folks do not display much patience with new users, but that's often due |
| 15:27.22 | ccvp | to their passion for the subject, not because they don't welcome your participation. |
| 15:27.24 | ccvp | oops. |
| 15:27.27 | Shotygun | No real-time, plain text |
| 15:27.36 | ccvp | I wonder if those that have those "attitudes" towards new users in here, do so because they are passionate, or because they really are actually tools :) |
| 15:27.49 | Shotygun | Based on what it assumes a sip device is busy? How can I check that list? |
| 15:28.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:28.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:28.14 | [TK]D-Fender | ccvp: Hey, I do not have an attitude problem, and I'll *KILL* anyone who says otherwise! |
| 15:28.19 | ccvp | :) |
| 15:28.21 | b11d` | :) |
| 15:28.24 | Wic3edOne | we are looking for a consultant to assist in design and implementation of a startup VOIP company down here in Florida if you have experience setting up systems that can support 5000 + VOIP customers we need you! |
| 15:28.32 | jblack | I'm having a problem with the polycom sp-301. I have set up 8NXXNXXXXXX (backward compatibility for humans) in my dialplan in a place the phones can reach. ATA devices can use the exten, but the polycoms aren't. |
| 15:28.47 | Faustov | [TK]D-Fender: WAIT! he failed! his phone led says he typed 8778, while i got here "878 no such extension" |
| 15:28.52 | Shotygun | putnopvut: Is there a way I can get a list of the devices it finds as "in use" so I can start debugging? |
| 15:28.57 | Faustov | [TK]D-Fender: so this means their dtmf is crap, right? |
| 15:28.57 | jblack | That leads me to think that polycoms have a list of valid dialing patterns. Is that the case? |
| 15:29.24 | ccvp | To the members of the expert community, who (correctly) perceive that Asterisk is like a web development language, this approach doesn't make any sense. To them, it's clear that you have to immerse yourself in Asterisk to appreciate its subtleties. Would one ask for a step-by-step guide to programming and expect to learn from it all that a language has to offer? |
| 15:29.57 | a-s | Maliuta: thanks |
| 15:30.08 | [TK]D-Fender | Faustov: It means "Holy shit, stop jumping to conclusions, and no 5-digit tests are NOT sufficient. Should have listen to that guy telling me I should do an EXHAUSTIVE test with an IVR as he suggested like 3 times already." |
| 15:30.32 | putnopvut | Shotygun: the "queue show" command will list all the queue members and their current device state. |
| 15:30.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:31.04 | [TK]D-Fender | jblack: Yup, the phone has its own dialplan. |
| 15:31.10 | ccvp | so when is the 1.6 book |
| 15:31.12 | ccvp | coming to B&N |
| 15:31.22 | Faustov | [TK]D-Fender: ok, sorry, i just didn't know "how exhaustive" |
| 15:31.24 | [TK]D-Fender | ccvp: So... when is 1.6? <- |
| 15:31.45 | Shotygun | putnopvut: Can I consider "sip show inuse" as a reliable source for indiciation for the queue? |
| 15:31.59 | ccvp | fender, why do you have an attitude with me? is it because you are passionate |
| 15:32.03 | ccvp | or is it other things :) |
| 15:32.08 | ccvp | heh |
| 15:32.12 | putnopvut | Shotygun: I actually didn't know that command existed :) . But if it does what I think it does, then yet. |
| 15:32.15 | putnopvut | s/yet/yes/ |
| 15:32.20 | [TK]D-Fender | Faustov: You're thinking asking 1 guy to enter a single number is good enough. That is jsut sad. If I had this problem, I'd have set the test ivr up and had him try at least a hundred DTMF rpesses to test. |
| 15:32.44 | Shotygun | putnopvut: Yeah I found it out accidently last night =) |
| 15:32.54 | Shotygun | Exists since 1.2 as far as I know |
| 15:32.55 | [TK]D-Fender | ccvp: Don't count your chickens before they're BBQ's :p |
| 15:32.59 | Faustov | [TK]D-Fender: but since he fails with 4 digits? |
| 15:33.24 | ccvp | or is it my nickname, that is silently and subliminally |
| 15:33.30 | ccvp | bringing up . . . 'cisco' . . . in here heh |
| 15:33.32 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 15:33.32 | [TK]D-Fender | Faustov: EXHAUSTIVE. I'm seriously not repeating myself again. |
| 15:33.32 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:34.50 | *** join/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-69560fba6fa46f45) |
| 15:34.56 | jblack | [TK]D-Fender: Ok, thanks |
| 15:35.28 | kensuke_ | Hi |
| 15:36.15 | outtolunc | breath in, breath out, breath in, hold it till you fall over <G> |
| 15:36.57 | Shotygun | putnopvut: ringinuse needs to be at the queue context and not global right? |
| 15:37.20 | putnopvut | Shotygun: that's correct. |
| 15:37.36 | Shotygun | I'll try it out again, I am reading the source I don't see anything wrong with it from first sight |
| 15:38.22 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 15:38.47 | Faustov | [TK]D-Fender: great, now i have to convince them they HAVE TO enter 100 digits :) |
| 15:39.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:40.45 | *** join/#asterisk Defraz (n=T0tal@65.39.96.74) |
| 15:43.38 | ccvp | outtolunc |
| 15:43.49 | ccvp | are there people saying you should goto a lamaz class because of weight issues? |
| 15:43.57 | ccvp | :) |
| 15:44.20 | outtolunc | eh? |
| 15:44.37 | ccvp | [10:36am] [outtolunc] breath in, breath out, breath in, hold it till you fall over <G> |
| 15:44.37 | outtolunc | i haven't been overweight for a few years now |
| 15:44.41 | outtolunc | ah |
| 15:44.54 | outtolunc | play on the 'exhaustive' part |
| 15:45.02 | Shotygun | I have a weird issue here. ringinuse does not work because it does not show in the 'show queue' that the channel is in use, it says not in use. Furthermore, when I do "sip show inuse" that sip device is not listed there at all. Along with many other sip devices which are not listed there. If it matters, qualify=yes and it's actively monitored successfully. |
| 15:45.10 | ccvp | That's one way to make people self concious about weight(men), tell them: hey are you going to a lamaz class? |
| 15:45.40 | Shotygun | Is there a chance the sip show inuse lists only those with call-limit ? |
| 15:45.46 | outtolunc | it made me think of beer-gut preggo look |
| 15:45.51 | ccvp | heh |
| 15:46.33 | outtolunc | actually seen a guy this weekend that looked that way.. not sure if it is a medical thing or not.. just strange |
| 15:46.54 | outtolunc | anyways <G> |
| 15:47.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:47.37 | *** join/#asterisk kripton1x (n=mike@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 15:47.48 | ccvp | well someone said some evil things about me |
| 15:47.56 | ccvp | so i said to him: hey, are you goint to a lamaz class |
| 15:48.01 | ccvp | and now he always looks at his stomach in a conversation, HEH |
| 15:48.23 | kripton1x | if I have exten => s,n,Macro(macroname), and macroname having a statement like SET(DN=${SIP_HEADER(TO):5} |
| 15:48.34 | kripton1x | would the macro recognize that variable that a normal context would? |
| 15:48.38 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:48.47 | outtolunc | say 'hey, aren't your shoes on the wrong feet?' <G> |
| 15:50.48 | [TK]D-Fender | kripton1x: All jsut dialplan. |
| 15:51.27 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 15:51.52 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:52.34 | spokra | any recomandations on a voip provider that allows asterisk and has flat rate incoming and outgoing? |
| 15:53.58 | *** join/#asterisk juice_d (n=juice_d@gozur.sunflowerbroadband.com) |
| 15:54.28 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:54.49 | *** join/#asterisk mbranca (n=matteo@mi-gw1.voismart.net) |
| 15:56.19 | *** part/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 15:56.35 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 15:57.17 | putnopvut | Shotygun: in order for a SIP device to be "in use" it has to have a call-limit specified. |
| 15:59.25 | Shotygun | putnopvut: Yeah I just found that out. Well apparently I think the whole story is related to this bug: http://bugs.digium.com/view.php?id=12249 |
| 15:59.37 | Shotygun | I think his issue is related to my mine |
| 15:59.45 | Shotygun | s/my// |
| 15:59.54 | Shotygun | jbot is cute =) |
| 16:00.58 | Faustov | [TK]D-Fender: 70-digit extension, entered twice, worked - does that give any idea? |
| 16:01.04 | [TK]D-Fender | ~areyouadog ? |
| 16:01.05 | jbot | Bark! Bark! |
| 16:01.18 | Faustov | heh :> |
| 16:01.42 | [TK]D-Fender | Faustov: including the # digits a pile of times? |
| 16:02.07 | Faustov | no |
| 16:02.25 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:02.39 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 16:03.38 | Faustov | retrying with a couple of # |
| 16:03.49 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
| 16:04.05 | [TK]D-Fender | Faustov: NO?!?! You've definitely killed your time quota today... |
| 16:04.33 | Faustov | [TK]D-Fender: heh, but wait, i just made them do it :> |
| 16:05.13 | Faustov | [TK]D-Fender: # wont let them enter more digits |
| 16:05.25 | Shotygun | There is something which is a bit confusing for me. Is it possible to use type=peer to both receive calls (like type=user) & engage calls and literally to avoid using type=user ? |
| 16:05.45 | Faustov | [TK]D-Fender: however works for me |
| 16:06.15 | [TK]D-Fender | Faustov: .... NO COMMENT |
| 16:06.40 | Faustov | [TK]D-Fender: no, please, comment this - what conclusion if only # fails for them? |
| 16:06.46 | [TK]D-Fender | Shotygun: Yes |
| 16:09.16 | Shotygun | For what do I need type=user then? or is it deprecated? |
| 16:09.34 | Shotygun | I missed that class I guess.. |
| 16:09.57 | [TK]D-Fender | Shotygun: somewhat. It has applicability for cases where inbound varies by host vs outbound, etc. |
| 16:11.47 | *** join/#asterisk mintee (n=mintee@72-165-177-94.dia.static.qwest.net) |
| 16:12.10 | *** join/#asterisk SplasPood (n=jwb@jwb.sh) |
| 16:16.20 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 16:25.23 | Uatec | dies at the invention of the word applicability |
| 16:25.47 | Qwell | invention? |
| 16:26.04 | Qwell | you do realize that it is a very real word already, right? |
| 16:26.32 | Uatec | it's not all THAT real is it? |
| 16:26.34 | cpm | it wasn't before it was invented |
| 16:26.39 | Uatec | it can't be more than 3% real |
| 16:26.48 | *** join/#asterisk FreedomBI (n=freedomb@mn01.freedombi.com) |
| 16:26.52 | Qwell | well, it's not applicabiliticism or anything |
| 16:27.18 | cpm | likes sewerage |
| 16:27.35 | cpm | thats my favorite one from last week. |
| 16:27.38 | Faustov | guys, if asterisk interprets all digits in an incoming call except the # key, does it mean there's an dtmf problem? |
| 16:28.05 | Qwell | there was a great coinage on the -users list about a week ago.. I wish I could remember what it was |
| 16:28.09 | cpm | every sentence I've read with that word, could have used sewage, with a slight change in wording, making it more clear. |
| 16:29.31 | Uatec | modern american is becoming more and more like newspeak |
| 16:29.31 | Uatec | why use the appropriate word when i can use another and just make it have some kind fo meaning by using suffixes or prefixes? |
| 16:29.57 | b11d` | its doubleplusgood |
| 16:30.03 | adyer | hi, I just started working with a Polycom IP320 - is it normal for it to take something like 3 mins to boot after starting to run sip.ld? |
| 16:30.06 | Uatec | it's doubleplusungood |
| 16:30.08 | Qwell | Uatec: would you prefer that we still used thee and thou? |
| 16:30.10 | Uatec | i mean it's fracking bad |
| 16:30.12 | b11d` | ;) |
| 16:30.20 | Qwell | language evolves |
| 16:30.23 | Uatec | Qwell, if appropriate, yes |
| 16:30.37 | Uatec | i know it does, that doesn't mean we should sacrifice meaning for the sake of change. |
| 16:30.50 | Qwell | verily it does |
| 16:30.55 | b11d` | lol |
| 16:31.00 | Qwell | doth. I fail. |
| 16:31.38 | Qwell | (I think that's the first time I've ever used 'verily' in a sentence) |
| 16:32.09 | caio1982 | Qwell: reprecation? |
| 16:32.25 | caio1982 | ah, that was on -dev |
| 16:33.08 | cpm | feels Uatec's pain, but there is no fix, so get over it. |
| 16:33.20 | Faustov | hmm could someone please help me with this issue? I understand this could be very basic, but if you at least pointed me to some manuals from which i could get a clear conclusion... |
| 16:34.02 | tzafrir | Fusoya, what type of phone? |
| 16:35.26 | tzafrir | Faithful, analog? SIP? |
| 16:35.43 | Qwell | caio1982: ahh, yeah, that's the one. "reprecation" |
| 16:35.55 | tzafrir | Fusoya, Faithful , sorry. The above two were for Faustov |
| 16:36.15 | Qwell | In response to a thread on reintroducing deprecated stuff - "Exactly what applications are being considered for reprecation?" |
| 16:36.39 | caio1982 | haha |
| 16:39.26 | Faustov | tzafrir: analog |
| 16:39.34 | [TK]D-Fender | Qwell: Oh, my $.02 on the "restoring deprecated dialplan apps" = NO. Move away from the past and let them mod their dialplans and read the friggen changelogs. |
| 16:39.42 | *** join/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
| 16:40.11 | tzafrir | Any chance that the '#' is considered to be a efature code? |
| 16:40.20 | Qwell | [TK]D-Fender: something like that |
| 16:40.49 | Faustov | tzafrir: how can i find out? |
| 16:41.40 | Faustov | tzafrir: # works when used in an incoming call from other places |
| 16:41.46 | Faustov | tzafrir: just fails when it comes from htem |
| 16:42.13 | tzafrir | analog phone? analog trunk? |
| 16:42.25 | *** join/#asterisk LuisTorres (n=chatzill@bl10-166-235.dsl.telepac.pt) |
| 16:42.44 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:42.56 | Faustov | tzafrir: analog phone with some dialplan |
| 16:46.15 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 16:46.15 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:46.32 | smps | hi, anyone from here had experience with grandstream HT386 and Fax over voip ? |
| 16:47.02 | lesouvage | maybe off topic but does "eth0: Oversized Ethernet frame cce9e1f0 vs cce9eqf0" ring a bell? |
| 16:47.35 | smps | ht386 was working with asterisk 1.2 very well, but when I updated to 1.4 i got problems with fax send/receive |
| 16:51.25 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 16:52.09 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:53.30 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 16:55.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 16:57.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.16 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:58.48 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-225.moldtelecom.md) |
| 17:06.45 | *** join/#asterisk golumn (n=acxty@201.220.132.138) |
| 17:06.59 | golumn | in which frequency shoul the gsm files be recorded? |
| 17:07.23 | tzafrir | What do you mean by "frequency"? |
| 17:07.40 | kotique | 8000 |
| 17:07.49 | golumn | kotique, 8MHz? |
| 17:07.56 | kotique | 8GHz |
| 17:08.21 | golumn | thanks |
| 17:08.33 | Strom_M | 8khz |
| 17:08.37 | Strom_M | dorks |
| 17:09.14 | golumn | thanks |
| 17:09.46 | kotique | in any case, it will error if it's not in right format |
| 17:10.05 | russellb | 8 GHz sample rate ... that's insane |
| 17:10.43 | kotique | it's not insane, it's the next generation super-fast speach recordign format |
| 17:10.45 | Strom_M | russellb: super high fidelity fart noises |
| 17:10.49 | russellb | heh |
| 17:11.01 | Strom_M | kotique: the ghz in this sense doesn't refer to clock speed |
| 17:11.11 | Strom_M | it refers to the number of samples per second |
| 17:11.13 | kotique | reall? |
| 17:11.14 | kotique | y |
| 17:11.21 | Strom_M | don't you know the nyquist theorem? |
| 17:11.33 | Qwell | ~nyquist theorem |
| 17:11.35 | kotique | nudist theorem ? |
| 17:11.37 | Qwell | ~nyquist |
| 17:11.40 | Qwell | silly bot |
| 17:11.45 | Strom_M | facepalms |
| 17:11.49 | Strom_M | go look it up on wikipedia |
| 17:11.52 | Strom_M | REQUIRED READING |
| 17:12.02 | Strom_M | points, threatens dunce cap |
| 17:12.04 | russellb | OMG how can you administer a phone system without knowing the technical details of digital audio processing?! |
| 17:12.11 | russellb | stares at Strom_M |
| 17:12.19 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 17:12.31 | kotique | yeah, how? |
| 17:13.09 | Strom_M | russellb: well you can, but that's a lot like maintaining a source tree and not knowing anything about C ;) |
| 17:13.36 | russellb | i don't know anything about C. |
| 17:13.49 | russellb | so what? |
| 17:14.01 | Strom_M | *shrug* |
| 17:14.02 | kotique | i know that C is worse than B |
| 17:14.03 | Qwell | russellb: you fake it well |
| 17:14.09 | *** part/#asterisk mintee (n=mintee@72-165-177-94.dia.static.qwest.net) |
| 17:14.09 | russellb | Qwell: thanks! |
| 17:14.16 | Strom_M | at least I still pwn you at bowling |
| 17:14.21 | russellb | true. :) |
| 17:14.22 | Strom_M | IIRC |
| 17:14.27 | russellb | probably |
| 17:14.35 | Qwell | Strom_M: I pwned em all at bowling too |
| 17:14.39 | Qwell | ..the second game |
| 17:14.45 | Qwell | first game was like an 80 >.< |
| 17:14.52 | Strom_M | heh |
| 17:15.24 | jjshoe | I don't think I've broken 80 in years :\ |
| 17:15.38 | jjshoe | although my best game ever was in the 220's |
| 17:16.48 | jjshoe | If only I bowled with the hand I masturbate with :\ |
| 17:16.54 | russellb | TMI |
| 17:17.05 | errr | lol |
| 17:17.09 | russellb | ~thwack jjshoe |
| 17:17.10 | jbot | ACTION beats jjshoe on the arm with a toaster |
| 17:17.18 | Strom_M | that counts as TMI? |
| 17:17.33 | Strom_M | is it shocking that people wank it in the privacy of their own homes? :) |
| 17:17.54 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:18.21 | rbd | hi, asterisk 1.4.x ...If I'm writing my own asterisk app that takes in a bunch of args (separated by |) that are all mandatory, can I still use AST_DECLARE_APP_ARGS to fetch them? I'm looking at the apps and they all use a bunch of different ways to get their args |
| 17:18.36 | jjshoe | Strom_M you'd think people would be thanking me for not dropping a link to a picture of me in the act? |
| 17:18.59 | Strom_M | *shrug* |
| 17:19.04 | jjshoe | :P |
| 17:19.05 | Strom_M | people are weird about that stuff |
| 17:19.58 | *** join/#asterisk s0lid (n=s0lid@58.69.102.130) |
| 17:20.49 | murdock_ut | So has a decision been made as to if asterisk is keeping the ZAP channel name? |
| 17:21.47 | murdock_ut | or is it going to be replaced with DAHDI/g0 |
| 17:22.06 | Strom_M | ? |
| 17:22.36 | murdock_ut | Strom_M: http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/ |
| 17:25.11 | [TK]D-Fender | Nice try that they went to make it an acronym, but they were clearly smoking the same crack as "trixbox" in trying to come up with the most ass-rammingly gay name possible. Seriously, wtf? |
| 17:25.42 | *** join/#asterisk TrentCreek (n=trent@cpe-70-117-198-98.rgv.res.rr.com) |
| 17:25.44 | outtolunc | blinks |
| 17:26.16 | *** join/#asterisk juice_d (n=juice_d@gozur.sunflowerbroadband.com) |
| 17:26.50 | [TK]D-Fender | (Not that have I have any problems with homophobia) |
| 17:26.55 | errr | I thought that was a joke post |
| 17:26.57 | outtolunc | myself i'd have thought they would call it something like Asterisk Hardware (plain and simple) and use 'Ast' as the tech |
| 17:27.43 | [TK]D-Fender | outtolunc: "TDM" would make a lot more sense. |
| 17:27.43 | sp00kz | instead its daddy |
| 17:27.58 | outtolunc | or that |
| 17:29.09 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:29.14 | [TK]D-Fender | With names like trixbox, dahdi, comedian mail, etc, how the hell is anyone supposed to take Asterisk seriously? |
| 17:29.37 | [TK]D-Fender | The joke is over already, so cut it out. |
| 17:29.55 | outtolunc | looks inside the box for the prize |
| 17:29.56 | creativx | dahdi da |
| 17:29.57 | creativx | sounds like |
| 17:30.03 | jameswf-home | I was thinking of posting a dead horse with a dahdi brand |
| 17:30.05 | creativx | childs play |
| 17:30.29 | [TK]D-Fender | jameswf-home: Flop that deceased equine! |
| 17:30.32 | errr | [TK]D-Fender: I agree |
| 17:30.33 | [TK]D-Fender | flog even! |
| 17:32.13 | Bananaskin | Hey guys looking a bit of advice, have in the past setup hylafax and external serial modem and had success both receiving and sending faxes. I have a TDM400P with 2 FXO modules and was thinking of adding an FXS module to to, is it possible to bridge a call if it is detected as a fax between the FXO and FXS ports (I have 2 pstn lines, both of which are connected to the TDM) |
| 17:32.54 | [TK]D-Fender | Bananaskin: forget FXS altogether and jsut use the IAXMode connetor to hylafax to let * support it directly |
| 17:33.01 | [TK]D-Fender | IAXModem* |
| 17:33.31 | [TK]D-Fender | Bananaskin: It uses the SpanDSP library to provide a virtual fax-modem to Hylafax. |
| 17:33.48 | errr | we use iaxmodem here with great success |
| 17:33.52 | Bananaskin | Would I not be correct in saying that faxing over VOIP is flaky and unreliable ? |
| 17:34.00 | [TK]D-Fender | Bananaskin: http://sourceforge.net/projects/iaxmodem |
| 17:34.20 | [TK]D-Fender | Bananaskin: When run on the * server itself its pretty reliable. |
| 17:34.40 | [TK]D-Fender | Bananaskin: Over anything more than a tightly controlled LAN, jsut forget it. |
| 17:34.46 | Bananaskin | Is the fax coming in over the internet or copper pair though ? |
| 17:34.58 | errr | [TK]D-Fender: do yall have any trouble with "Brother" fax machines using your iaxmodem + hylafax setup? |
| 17:35.00 | [TK]D-Fender | Bananaskin: it comes in however your call comes into * |
| 17:35.17 | Bananaskin | Would it support over the TDM ? |
| 17:35.27 | [TK]D-Fender | errr: I just installed it at home over the weekend, and at work this morning. Haven't tested personally. |
| 17:35.30 | [TK]D-Fender | Bananaskin: Yup |
| 17:35.36 | Bananaskin | hmmm |
| 17:35.45 | [TK]D-Fender | Bananaskin: So call comes in on your TMD card and Hylafax can process it right off there. |
| 17:36.05 | errr | [TK]D-Fender: ah ok, we have had it in production here for over a year, the only problems we seem to have are with certian brother fax machines |
| 17:36.12 | Bananaskin | Have you seen PIKAFAX from PIKA Technologies |
| 17:36.40 | Bananaskin | module for asterisk T30 faxing using exiting hardware |
| 17:36.47 | [TK]D-Fender | Bananaskin: read something about them a few odd months ago. Fell into my "thats nice" category as something I'm not likely to look at twice... |
| 17:37.17 | Bananaskin | k |
| 17:39.30 | Bananaskin | [TK]D-Fender, does IAXmodem support the high res faxes ? |
| 17:40.29 | [TK]D-Fender | Bananaskin: I think so, you'd have to go look. |
| 17:40.51 | jameswf-home | jbot dahdijoke is http://files.jameswf.info/dahdihorse.jpg |
| 17:40.51 | jbot | okay, jameswf-home |
| 17:42.21 | tzanger | I don't get this dahdi thing |
| 17:43.35 | *** join/#asterisk mintee (n=mintee@72-165-177-94.dia.static.qwest.net) |
| 17:44.03 | mintee | how do you test is an extension exists? For instance if a SIP phone is actually online. |
| 17:45.31 | Bananaskin | only posts that I see in the IAXModem forums is ones of - errors and help cant get it to work. |
| 17:45.47 | mintee | Bananaskin, what's the problem? |
| 17:46.11 | jameswf-home | ~dahdi |
| 17:46.11 | jbot | [dahdi] Digium/Asterisk Hardware Device Interface. The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi |
| 17:46.43 | errr | mintee: if you are using the manager you can find out with ExtensionState |
| 17:47.32 | [TK]D-Fender | Bananaskin: I highly recommend you just try it... |
| 17:47.56 | Strom_M | ABE C is based on Asterisk 1.4, right? |
| 17:50.01 | russellb | yes |
| 17:50.11 | Strom_M | ok |
| 17:50.25 | Strom_M | I have a client who keeps going on about "asterisk business edition 1.6" |
| 17:50.38 | Strom_M | just wanted to make sure I wasn't going mad :) |
| 17:50.44 | russellb | heh |
| 17:51.19 | golumn | hey guys, how can I reproduce gsm files on linux? |
| 17:51.21 | [TK]D-Fender | Strom_M: Is that compatible with his Windows 7 Virtual SOHO Workstation Cluster? |
| 17:51.27 | golumn | *play |
| 17:51.29 | [TK]D-Fender | golumn: "cp" |
| 17:51.46 | golumn | *by reproduce I mean play |
| 17:51.50 | golumn | ;) |
| 17:52.12 | errr | doesnt play play them? |
| 17:52.48 | golumn | thanks |
| 17:52.58 | [TK]D-Fender | golumn: or VLC |
| 17:54.43 | TrentCreek | Fender must be retired |
| 17:55.05 | TrentCreek | on here 34/7 |
| 17:55.35 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 17:57.46 | [TK]D-Fender | TrentCreek: At least MY math skills haven't eroded ;) |
| 17:58.08 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 17:58.10 | Strom_M | I think he means 34/9 |
| 17:58.47 | jameswf-home | jbot: 43/9 |
| 17:58.47 | jbot | 4.777777777778 |
| 17:58.54 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-164-254.red.bezeqint.net) |
| 17:58.59 | jameswf-home | jbot: 34/7 |
| 17:58.59 | jbot | 4.857142857143 |
| 17:59.16 | Dovid | can anyone reccomend a 2 port FXO ATA |
| 17:59.24 | rob0 | let's see him divide by zero |
| 17:59.24 | Dovid | jbot: 100/100 |
| 17:59.25 | jbot | 1 |
| 17:59.29 | Dovid | hehe |
| 17:59.35 | jameswf-home | jbot: 10000000/0 |
| 17:59.45 | jameswf-home | crash lol |
| 17:59.45 | rob0 | lol! |
| 18:00.18 | Dovid | i love the bots that copy what u say. all u do is /quit and they follow u ;) |
| 18:00.33 | Dovid | jbot: 100/100 |
| 18:00.34 | jbot | 1 |
| 18:00.48 | Dovid | jbot: say hi |
| 18:00.49 | jbot | hi |
| 18:00.55 | Dovid | jbot: say foo |
| 18:00.56 | jbot | foo |
| 18:01.01 | Dovid | jbot: say /quit |
| 18:01.02 | jbot | /quit |
| 18:01.05 | Dovid | dman it |
| 18:01.09 | jameswf-home | hah hah |
| 18:01.22 | jjshoe | hrm does anyone kick for colors? |
| 18:01.28 | jjshoe | make jbot get kicked for spewing back colors :D |
| 18:01.32 | jameswf-home | jbot: dropdatabase; |
| 18:01.32 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 18:01.43 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:01.48 | SQLDarkly | Does anyone know of a usb audio splitter? |
| 18:01.55 | Dovid | teeheee |
| 18:01.58 | rob0 | ~botsnack |
| 18:01.58 | jbot | aw, gee, rob0 |
| 18:02.04 | [TK]D-Fender | Dovid: spa-2102 |
| 18:02.10 | Dovid | thanks TK |
| 18:02.23 | [TK]D-Fender | SQLDarkly: THAT PRODUCT DOESN'T MAKE ANY SENSE. |
| 18:02.44 | SQLDarkly | The ability to Y-Jack makes perfect sense |
| 18:02.49 | Dovid | TK: http://www.powerbuysdirect.com/ssproduct.asp?pf_id=11894311 |
| 18:02.52 | Dovid | seems to have 2 fxs |
| 18:02.56 | rob0 | Isn't 2102 dual FXS? |
| 18:03.24 | [TK]D-Fender | Dovid: # 2 x RJ-11 FXS <- |
| 18:03.24 | SQLDarkly | training purposes to have an agent plug in a 2nd headset and listen when rdy unmute and try a call for yourself |
| 18:04.07 | *** join/#asterisk jrprado (n=jrprado@189.58.27.150.adsl.gvt.net.br) |
| 18:04.09 | [TK]D-Fender | SQLDarkly: Never heard of such a product, nor can I imagine anyone making one. No viable market for it. |
| 18:04.30 | keith4 | usb... audio splitter? |
| 18:04.34 | [TK]D-Fender | Dovid: taht site sucks, and its still clarified further down. use your eyes. |
| 18:04.37 | keith4 | that doesn't make sense on several levels |
| 18:04.47 | SQLDarkly | yes like a traditional platronics headsets |
| 18:04.57 | [TK]D-Fender | keith4: Would be the first request like this he's made in here either ;) |
| 18:05.01 | SQLDarkly | linux softphone i forget the name allows multiple devices |
| 18:05.38 | [TK]D-Fender | SQLDarkly: yes well plantronics SELLS tons of their stuff, and thats just an audio cable, not a "bridged soundcard that will miraculously merge with ANOThER unit on the same PC. |
| 18:05.42 | keith4 | why wouldn't you just do the splitting in analog, if you insist on splitting for some reason? |
| 18:05.57 | SQLDarkly | Your thinking about it to complex |
| 18:06.15 | SQLDarkly | heh they can do it with softphones via linux distro suse i believe |
| 18:06.16 | [TK]D-Fender | SQLDarkly: A dumb audio splitter is too complx? |
| 18:06.22 | *** join/#asterisk mattchis (n=IceChat7@adsl-99-130-234-246.dsl.hstntx.sbcglobal.net) |
| 18:06.22 | keith4 | *You're*\ not being clear |
| 18:06.32 | TrentCreek | what's what with 34 hours a day? |
| 18:06.33 | keith4 | better yet, why not use another phone, and listen in to the call? |
| 18:06.36 | Dovid | nexpensive, easy to install and simple-to-use, the Linksys SPA2102 connects a standard telephone or fax machine to IP-based data networks. VoIP service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet. |
| 18:06.44 | *** join/#asterisk hsv-AL (n=ccvp@66.0.46.210) |
| 18:06.47 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-46-230.austin.res.rr.com) |
| 18:06.53 | Dovid | from voip supply. not hona bust ur balls. u have helped me a lot |
| 18:07.10 | hsv-AL | Dovid |
| 18:07.18 | hsv-AL | barook atah adonai elehenu melech halum |
| 18:07.26 | [TK]D-Fender | Dovid: And what is that incomplete and broken paste supposed to tell me? |
| 18:07.34 | Dovid | TK: nm |
| 18:07.41 | keith4 | [TK]D-Fender: what an ATA is? ;-) |
| 18:07.43 | Dovid | hsv-AL: Congrats you know hebrew |
| 18:07.46 | Dovid | !ATA |
| 18:07.56 | sp00kz | ~ata |
| 18:07.57 | jbot | from memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:08.00 | mattchis | Can anyone tell me of a way to see if asterisk 1.4.18 is causing a memory leak on my server? |
| 18:08.11 | [TK]D-Fender | Dovid: http://www.voipsupply.com/index.php?cPath=96_118 |
| 18:08.24 | [TK]D-Fender | Dovid: Notice what page this item appears on? |
| 18:08.47 | [TK]D-Fender | Dovid: Notice the day-glow neon big print in its description "Linksys SPA2102 NA Analog VoIP Adapter (2) FXS, (2) RJ45" |
| 18:08.53 | Dovid | FXS/FXO page ? |
| 18:09.11 | ReDNeQ | how can i diagnose static/noise on zaptel channel |
| 18:09.18 | [TK]D-Fender | Dovid: http://www.voipsupply.com/index.php?cPath=96_118 <- Read. The. Page's. TITLE <---- |
| 18:09.41 | Dovid | FXS FXO SIP |
| 18:09.44 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 18:10.25 | FarrisG | Suddenly ChanSpy is acting screwy on me. When I try to switch channels, it beeps as it used to, but then just repeats the same channel and does not change. Any idea what the deal is? |
| 18:10.26 | [TK]D-Fender | Dovid: I swear you must have 2 gaping holes where eyes should be. |
| 18:10.44 | hsv-AL | is it possible to emulate? FXO/FXS ports in software, sort of how you cant emulate the switch asics |
| 18:10.45 | keith4 | that must look creepy |
| 18:10.45 | hsv-AL | in dynamips? |
| 18:10.49 | Dovid | Dual FXS Adaptors |
| 18:10.54 | Dovid | and i need DUAL FXO !!!!!!!!!!!!!!!!!!! |
| 18:11.10 | Dovid | TK: In regards to my eyes u may be correct. it has been a long day |
| 18:11.16 | [TK]D-Fender | Dovid: Ah well, thats another matter :p |
| 18:11.31 | Qwell | hsv-AL: how do you propose handling voltage and layer 1 connectivity, in software? |
| 18:11.32 | Dovid | [20:59] <Dovid> can anyone reccomend a 2 port FXO ATA |
| 18:11.40 | [TK]D-Fender | Dovid: 2-port FXO? Nothing normal out there really.. the rest art at 4 ports. |
| 18:11.40 | Dovid | and here TK I thought you were the crazy one |
| 18:12.14 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-230-58.unitz.ca) |
| 18:12.18 | hsv-AL | qwell |
| 18:12.23 | [TK]D-Fender | Dovid: yeah we each made a little mix-up there. |
| 18:12.26 | Dovid | they are nice and costly. prob. gona get 2 single fxo's. goto luv cheap clients whom I HATE dealing with (but the boss insists) |
| 18:12.27 | hsv-AL | lets emulate layers of oxide while were at it |
| 18:12.28 | hsv-AL | :) |
| 18:12.32 | mintee | errr, yeah, i'm not using the manager... |
| 18:12.48 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:13.26 | Dovid | TK: thaks for trying to help |
| 18:13.40 | [TK]D-Fender | Dovid: Yeah, sorry that took a left turn :) |
| 18:13.55 | [TK]D-Fender | Dovid: If you're cheap, 2x SPA-3102's |
| 18:13.57 | hsv-AL | qwell, im just looking into messing with call manager within dynamips |
| 18:14.06 | hsv-AL | so i was on a thought tangent of fxo/fso in software heh |
| 18:14.52 | Dovid | TK: Thats what I was thinking |
| 18:15.41 | Qwell | hsv-AL: well, you'll have to get the source for call manager, add a channel driver for it, and...oh, wait, you can't |
| 18:15.53 | hsv-AL | :) |
| 18:17.01 | russellb | hsv-AL: what's with the name? |
| 18:17.29 | hsv-AL | ? |
| 18:17.29 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:17.34 | hsv-AL | of dynamips? |
| 18:17.35 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 18:17.35 | russellb | what does it stand for? |
| 18:17.39 | hsv-AL | oh, its where i live |
| 18:17.45 | russellb | Huntsville, Alabama? |
| 18:17.46 | hsv-AL | yah |
| 18:17.57 | russellb | orly |
| 18:18.01 | talntid | goes into stalker mode |
| 18:18.03 | hsv-AL | ? |
| 18:18.12 | russellb | knows of a company related to Asterisk that also is there ... |
| 18:18.21 | sp00kz | lol |
| 18:18.28 | tzafrir | And I thought people from there were banned here |
| 18:18.31 | hsv-AL | yeah i know, its like 15min away in RP |
| 18:18.43 | hsv-AL | Im off drake by parkplace mall |
| 18:18.44 | tzafrir | ducks |
| 18:18.52 | russellb | ah, ok, cool |
| 18:18.54 | [TK]D-Fender | preps another ICMB |
| 18:19.01 | talntid | Fender |
| 18:19.06 | Qwell | I once took the Drake exit off the parkway. |
| 18:19.08 | talntid | I worked on ICBM's for 3 years |
| 18:19.08 | Qwell | once. |
| 18:19.08 | russellb | i work for digium which is why i asked ... |
| 18:19.21 | russellb | waves in the SE direction |
| 18:19.24 | hsv-AL | i see, well drake is a sorta trashy area, but theres a complex of businesses here |
| 18:19.26 | hardwire | I was a sasquatch/yeti hunter. |
| 18:19.26 | sp00kz | never made that mistake twice? |
| 18:19.33 | hsv-AL | near operon tech |
| 18:19.37 | [TK]D-Fender | talntid: add AMEX, VISA, IHOP, and TCBY FTW! |
| 18:19.44 | talntid | heh. ;) |
| 18:19.54 | hsv-AL | i thought you were going to kick me, because i was talking about dynamips |
| 18:20.15 | rob0 | I get glasses out thattaway. |
| 18:20.27 | *** join/#asterisk laurier57 (n=drj33@gateway.centerpointe.com) |
| 18:20.29 | russellb | don't even know what that is. |
| 18:20.41 | hsv-AL | can pop ios images in it |
| 18:20.46 | hsv-AL | no need to by c hardware |
| 18:20.51 | rob0 | Drake & South Memorial Parkway |
| 18:20.53 | Dovid | besides for Digium isnt the rest of AL hicks ? well thats atleast what they teach us on the east coast (NY/NJ) |
| 18:21.03 | rob0 | yes |
| 18:21.10 | hsv-AL | rob0, where you at? |
| 18:21.11 | russellb | s/Digium/Huntsville/ |
| 18:21.13 | russellb | then ... pretty much |
| 18:21.21 | rob0 | well, digium and a few other places in HSV |
| 18:21.38 | russellb | HSV is nice, though ... |
| 18:21.44 | hsv-AL | i should assume AL as well "tux" a loosa, open source jab at |
| 18:21.48 | hsv-AL | tuscaloosa |
| 18:21.58 | hsv-AL | hence your one of us |
| 18:22.11 | rob0 | actually good guess, but no, my sweet home alabammy is closer to Florence. |
| 18:22.22 | hsv-AL | Killen = junk |
| 18:22.33 | hsv-AL | good noodling in Flor, grab them catfish by the pond |
| 18:22.39 | rob0 | yeah not far from Killen too. |
| 18:23.03 | jjshoe | thank god I'm not a country boy |
| 18:23.08 | Qwell | rob0: glasses? Sullen's? |
| 18:23.16 | rob0 | America's Best |
| 18:23.18 | Qwell | Doc Sullen is the man :p |
| 18:23.20 | hsv-AL | jjshoe, im not originally from here |
| 18:23.24 | jbeez | America's Worst |
| 18:23.25 | hsv-AL | i grew up in NJ, moved here for work heh |
| 18:23.32 | rob0 | haha but they're cheap |
| 18:23.34 | hsv-AL | been here for 2yrs |
| 18:23.54 | jbeez | I went there, it was a disaster, then when I wanted to know my contact lens prescription like a year and a half later they refused to tell me |
| 18:24.01 | hsv-AL | I started laughing alot when people would come into work like 4 hours late, because it "flurried" outside |
| 18:24.03 | Dovid | NJ rocks but NY is better |
| 18:24.11 | jbeez | so I called their corp number and complained it was a hipaa violation to withhold my medical records from me |
| 18:24.11 | hsv-AL | and all the residents stockpiled on milk/eggs/water as if a nuclear bomb went off |
| 18:24.23 | jjshoe | jbeez how well did that work out for you? |
| 18:24.26 | jbeez | the manager called me back and told me I could pick it up anytime, it would be available at the end of the day |
| 18:25.23 | jbeez | the same manager I fought with on the phone for 1/2 hr trying to get the info because I was trying to order contacts online, I could have just waited till I got home and checked the box, but I don't like being lied to so I had to break some balls |
| 18:26.16 | jbeez | I even found the section of the hipaa guidlines for them that says they must disclose this information to me if I request it |
| 18:29.14 | *** part/#asterisk Dovid (n=Dovid@bzq-79-177-164-254.red.bezeqint.net) |
| 18:34.34 | *** join/#asterisk ManxPower (n=manxpowe@12.sub-70-220-153.myvzw.com) |
| 18:35.11 | *** part/#asterisk adyer (n=adyer@adsl-69-223-13-78.dsl.chcgil.ameritech.net) |
| 18:40.54 | *** join/#asterisk DrkShadow_ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 18:41.44 | DrkShadow_ | Is there a way to catch calls that go to an invalid context? I'm using a GoSub, but if the context doesn't exist, it doesn't return -- the channel is disconnected. |
| 18:42.51 | ManxPower | DrkShadow_: What technology? |
| 18:43.05 | *** join/#asterisk hsv-AL (n=ccvp@66.0.46.210) |
| 18:43.42 | *** join/#asterisk bkw___ (n=brian@adsl-71-153-169-69.dsl.tul2ok.sbcglobal.net) |
| 18:44.15 | ManxPower | I doubt you can do what you want to do, since calls should never ever land in an invalid context. |
| 18:44.34 | jeffspeff2 | anybody know off the top of their head what the max bitrate is in * for a .wav file? I'm getting an unexpected bitrate error on a file that I made. |
| 18:44.49 | ManxPower | 64Kbps |
| 18:45.19 | jeffspeff2 | heres the error |
| 18:45.21 | jeffspeff2 | format_wav.c:156 check_header: Unexpected freqency 44100 |
| 18:45.22 | ManxPower | or if you are talking just about .wav file I think it's 8Khz |
| 18:45.33 | jeffspeff2 | oh, ok |
| 18:45.42 | jeffspeff2 | thanks ManxPower |
| 18:45.48 | ManxPower | *nod* make it frequency 8192 |
| 18:45.52 | Qwell | O.o |
| 18:45.58 | ManxPower | or maybe 8000, I don't recall which is correct |
| 18:45.59 | Qwell | 8000 |
| 18:46.13 | jeffspeff2 | ok |
| 18:46.40 | DrkShadow_ | manx: ... the Asterisk dialplan... |
| 18:47.07 | DrkShadow_ | call comes in via sip, goes through the dialplan, if the dialplan sends it to an invalid extension I want to be able to recover from it in a specific way |
| 18:47.23 | Qwell | DrkShadow_: in the question above, the technology is sip |
| 18:47.33 | codefreeze-lap | DrkShadow_: how about defining the 'i' extension? |
| 18:47.47 | DrkShadow_ | code: in every non-existant section possible? |
| 18:47.56 | codefreeze-lap | (invalid) |
| 18:48.09 | codefreeze-lap | well, at least, in all the contexts that might be involved |
| 18:48.15 | DrkShadow_ | the context I'm coming _from_ has an i extension. The context I aws going to doesn't exist. |
| 18:48.35 | DrkShadow_ | simple capitalization error, but I'd like to catch it in the future |
| 18:48.35 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:48.50 | codefreeze-lap | DrkShadow_: I can see an extension being iffy-- but not the context. |
| 18:49.01 | ManxPower | DrkShadow_: Make up your mind. Do you want to catch invalid extensions or invalid contexts? One is easy, one may not be possible. |
| 18:49.02 | *** join/#asterisk twitchnln (n=twitch@c-76-105-88-129.hsd1.ga.comcast.net) |
| 18:49.11 | twitchnln | good afternoon |
| 18:49.16 | ManxPower | DrkShadow_: The i extension is only for IVRs |
| 18:50.05 | ManxPower | DrkShadow_: I use wildcard/pattern matches to catch invalid extensions not from an IVR |
| 18:50.20 | ManxPower | exten => _X.,1,Playback(invalid-extension-you-moron) |
| 18:53.09 | DrkShadow_ | btw, i extension works in _any_ context |
| 18:53.45 | ManxPower | DrkShadow_: i will not be matched in non-include =>'d contexts |
| 18:53.49 | ManxPower | and it only works for IVRs |
| 18:54.24 | keith4 | for some values of "works" |
| 18:54.50 | ManxPower | keith4: and for some values of "any" |
| 18:54.56 | keith4 | grins |
| 18:55.06 | ManxPower | He didn't answer my question so I'm not helping him further |
| 18:55.06 | DrkShadow_ | i does not work only for IVR's. |
| 18:55.13 | keith4 | it would be neat to have a globally-defined invalid handler |
| 18:55.13 | rob0 | And if you include => the context with exten => _X. last, it is the last thing it tries to match? |
| 18:55.26 | *** join/#asterisk chadmaynard (n=chadm@pdpc/supporter/active/chadmaynard) |
| 18:55.26 | keith4 | and then it could be scoped, so you could override it in IVR contexts, for example |
| 18:55.48 | ManxPower | keith4: I make sure no calls ever land in a user supplied context |
| 18:55.56 | rob0 | I take it that exten => _X. has to be in a context by itself? |
| 18:56.03 | ManxPower | rob0: no, it does not. |
| 18:56.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.31 | keith4 | ManxPower: "user supplied"? |
| 18:56.32 | ManxPower | DrkShadow_: Best of luck with that. |
| 18:56.37 | rob0 | it's matched in the order given in extentions.conf? |
| 18:56.57 | keith4 | it's matched in the order of... most specific match first, no? |
| 18:57.18 | rob0 | this bit has confused me before |
| 18:57.19 | ManxPower | rob0: like everything else in the dialplan it's matched to the most specific pattern/extension in the current context, then it searches included contexts, just like everything else in the dialplan |
| 18:57.25 | keith4 | thus, _X. is a catch-all for stuff that didn't match *anything* else |
| 18:57.54 | rob0 | ok, but if _X. is in the current context, includes are ignored, right? |
| 18:58.12 | ManxPower | keith4: Actually it's a catch-all for numbers of 1 digit followed by any digits or characters |
| 18:58.19 | ManxPower | rob0: Why in the WORLD you say that? |
| 18:58.39 | rob0 | um, I thought that's what you said :) |
| 18:58.46 | ManxPower | yes, if the included context ALSO had an _X. then the one in the current context is matched first. |
| 18:59.07 | keith4 | ManxPower: well, yes. so, it doesn't help people who make up *XX combinations... |
| 18:59.15 | rob0 | but more specific patterns in included contexts are checked first? |
| 18:59.18 | ManxPower | BTW, _X. is just a pattern match, nothing else, nothing special |
| 18:59.53 | ManxPower | rob0: Most specfic extension matching the dialed digits will match in the current context, if no match is found it will search the include'd contexts. |
| 19:00.06 | ManxPower | keith4: Huh? |
| 19:00.35 | keith4 | eh, forget it |
| 19:03.34 | rob0 | thanks, I'll try some things. |
| 19:06.39 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 19:08.49 | cpm | gives rob0 some things to try |
| 19:13.37 | *** join/#asterisk kannan (n=chatzill@123.201.60.110) |
| 19:13.55 | *** part/#asterisk bkw_ (n=brian@adsl-71-153-169-69.dsl.tul2ok.sbcglobal.net) |
| 19:13.55 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 19:14.12 | muiro | is there a way to see what module provides an application in asterisk? |
| 19:14.38 | kannan | ManxPower , hi, thanks for the help, it was a telco issue as you said, and the TE410P is working ok now |
| 19:15.48 | *** join/#asterisk bkw___ (n=brian@68-240-166-25.area5.spcsdns.net) |
| 19:21.36 | *** join/#asterisk robevans (n=robevans@OL6-231.fibertel.com.ar) |
| 19:24.18 | *** join/#asterisk docelmo (n=chatzill@206.248.239.194) |
| 19:24.41 | docelmo | Say anyone ever hear of asterisk making sound files come out all muffled when played? Do you know what causes this and how to fix it? |
| 19:24.43 | *** join/#asterisk mmurdock (n=chatzill@70.99.184.194) |
| 19:28.54 | *** join/#asterisk bkw___ (n=brian@adsl-71-153-169-69.dsl.tul2ok.sbcglobal.net) |
| 19:29.21 | docelmo | uhh anyone alive? |
| 19:30.21 | tzafrir | nobody's alive |
| 19:30.32 | mmurdock | Not here. |
| 19:30.51 | *** join/#asterisk lwizardl (n=jamesdav@adsl-64-149-133-14.dsl.klmzmi.sbcglobal.net) |
| 19:30.57 | Corydon76-dig | Nobody here but us zombies |
| 19:31.02 | lwizardl | hi |
| 19:31.20 | Corydon76-dig | BRAAAAAAAAAAAAINS |
| 19:31.21 | *** part/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 19:31.23 | hardwire | hi zombies |
| 19:31.26 | lwizardl | I was wondering if asterisk would work for my needs |
| 19:31.29 | mmurdock | mwaaaaa |
| 19:31.33 | hardwire | lwizardl: most likely |
| 19:31.39 | outtolunc | loads the sniper rifle |
| 19:31.42 | mmurdock | lwizardl: :Yup, no problem. |
| 19:31.46 | djs26 | shakes magic 8 ball |
| 19:31.51 | hardwire | lwizardl: undoubtedly |
| 19:31.57 | Corydon76-dig | You have needs? |
| 19:32.18 | hsv-AL | heh |
| 19:32.31 | lwizardl | I want to have a setup where I have 2 voip numbers, 1 for Voip routed to regular phone, 1 for voip routed to my fax machine. |
| 19:32.50 | Corydon76-dig | ixnay on the voip routed to your fax machine |
| 19:32.57 | hardwire | lwizardl: actually, it won't work |
| 19:33.04 | hardwire | haw haw haw |
| 19:33.11 | hardwire | we were all SO SURE HE WASN'T INSANE |
| 19:33.16 | Corydon76-dig | See "latency" |
| 19:33.17 | lwizardl | but I want to have it save all calls into data files such as mp3/pdf for example |
| 19:33.29 | hardwire | lwizardl: your fax machine does all that? |
| 19:33.35 | hardwire | so does hylafax |
| 19:33.36 | *** join/#asterisk marc7 (n=marc@p236-226.public.uvic.ca) |
| 19:33.40 | hardwire | and iaxmodem |
| 19:33.41 | hsv-AL | lwizardl, aren't you the one that was talking about A not having 'on the fly' sha1/aes encoded traffic? |
| 19:34.01 | marc7 | If I'm seeing the following being flooded (once a second) in my asterisk log... should I be suspecting foul play? |
| 19:34.01 | marc7 | <PROTECTED> |
| 19:34.02 | marc7 | <PROTECTED> |
| 19:34.06 | lwizardl | hardwire: no but I figured there might be a linux way to do it |
| 19:34.12 | hardwire | marc7: yes |
| 19:34.13 | lwizardl | hsv-AL: nope first time here |
| 19:34.18 | hardwire | lwizardl: there is. |
| 19:34.20 | marc7 | I'm not the one attaching/detaching the console |
| 19:34.35 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 19:34.46 | hardwire | marc7: maybe you are running something like asterisknow or something? |
| 19:34.48 | marc7 | hardwire: shouldn't the asterisk console only be accessible by local sockets. |
| 19:34.57 | hardwire | there are daemon processes that connect to the console, etc.. |
| 19:34.58 | marc7 | ? |
| 19:35.07 | marc7 | this is obviously not one of those. |
| 19:35.09 | hardwire | what OS/dist/etc.. |
| 19:35.25 | marc7 | debian-etch, 1.4.18 (maybe not the most current asterisk build) |
| 19:35.29 | TrentCreek | windows vista |
| 19:35.44 | hardwire | TrentCreek: cute |
| 19:35.53 | hardwire | marc7: that your own backport? |
| 19:36.02 | TrentCreek | buwahahaha |
| 19:36.02 | marc7 | hardwire: compiled via src |
| 19:36.06 | hardwire | marc7: gotcha |
| 19:36.14 | marc7 | i'll update to latest asterisk |
| 19:36.20 | hardwire | marc7: why? |
| 19:36.22 | marc7 | but I don't understand what's connecting, and how I would block it |
| 19:36.28 | hardwire | oh |
| 19:36.36 | hardwire | well that's sorta simple |
| 19:36.57 | hardwire | what other processes are running? |
| 19:36.57 | adorah | Hi does anyone here have experience with configuring polycom 501? |
| 19:37.02 | *** join/#asterisk gardo (n=gardo@121.97.109.122) |
| 19:37.07 | muiro | any way to tell what module is providing a given application? |
| 19:37.08 | hardwire | what do you have in your crontabs? |
| 19:37.16 | hardwire | muiro: voip-info.org knows all |
| 19:38.13 | muiro | see, I don't think that will work. What I know is the application, what I don't know is the module |
| 19:38.29 | marc7 | hardwire: nothing in crontab would run this frequently |
| 19:38.30 | hardwire | strings /var/lib/asterisk/modules/* | grep :) |
| 19:39.13 | marc7 | hardwire: it's entirely possible that it was due to two instances of safe_asterisk running |
| 19:39.15 | muiro | right right |
| 19:39.16 | muiro | thanks |
| 19:39.29 | marc7 | looks like it. |
| 19:39.33 | lwizardl | hardwire: so how would i get an asterisk box configured for my needs? |
| 19:40.05 | TrentCreek | ~book |
| 19:40.05 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:40.10 | hardwire | lwizardl: install linux, install hylafax, install iaxmodem, install asterisk, then fold-in lots of time. |
| 19:40.13 | kannan | has any one use voxee fir sip termination to usa? |
| 19:41.30 | hardwire | ~buybook |
| 19:41.31 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 19:41.33 | hardwire | :) |
| 19:42.09 | keith4 | faxing over VoIP = ANGRY FACE |
| 19:42.10 | hardwire | it works, just not well enough to be supportable. |
| 19:42.35 | hardwire | on a super low latency network with no isses at all on an ideal day. no sun spots. |
| 19:43.11 | hardwire | keith4: turn t38modem h323 product into iax/socket |
| 19:43.22 | CunningPike | Does anyone have any opinions on SIPx vs. SER for UDP to TCP conversion? |
| 19:43.22 | lwizardl | well I used to have vonage and had a voice line and a fax data line and that worked great. so I'm sure linux could do the same probably with better results |
| 19:43.37 | hardwire | lwizardl: that's a bit different. |
| 19:44.02 | hardwire | vonage actually intercepts the fax calls then replays them to you. |
| 19:44.07 | hardwire | similarly to hylafax |
| 19:44.22 | keith4 | does it really? |
| 19:44.24 | keith4 | wow |
| 19:44.26 | lesouvage | Does anybody know a good starting point for the combination Asterisk and OpenSer. On voip-wiki there is an instruction but it seems to be for older version . |
| 19:44.34 | hardwire | I'd really just appreciate everybody more if they stopped faxing all together. |
| 19:44.44 | hardwire | it's just wrong people |
| 19:44.46 | hardwire | its wrong |
| 19:44.59 | hardwire | and you're all crazy for every wanting to use it ever. tell everybody you know. |
| 19:45.11 | hardwire | LETS GET THE WORD OUT THAT SCAN -> EMAIL = GLORY |
| 19:45.32 | hardwire | LETS GET THE WORD OUT THAT SOURCE DOCUMENT -> EMAIL = AWESOME |
| 19:45.39 | hardwire | sigh. |
| 19:45.40 | JenniferAkemi | why do you want to use openser lesouvage? (not saying it's not a good idea, just wondering what it's for) |
| 19:45.44 | hardwire | I'm going to lunch |
| 19:45.56 | lesouvage | hardwire: costumers want it. You can not simply refuse the wish of te costumer. |
| 19:46.20 | hardwire | lesouvage: it's all about changing what the customer wants |
| 19:46.21 | hardwire | ask Nike |
| 19:47.02 | hardwire | the less we give in to customers the more we give to better standards developed with tech in mind. :) |
| 19:47.08 | hardwire | tootles.. lunch.. my blood sugar are off. |
| 19:47.25 | znoG_ | hey, if I wanted to write a page in PHP to find out different things about Asterisk (registered SIP/IAX clients, etc) .. anyone know of a PHP class that does the hard work? |
| 19:47.40 | docelmo | znoG_: PHPAGI |
| 19:47.46 | hardwire | FTW |
| 19:47.47 | lesouvage | JenniferAkemi: For now just to play around but for serious implementation it seem to be wise to use OpenSer and use Asterisk as one of the SIP implementation. For me it is the next frontier. |
| 19:47.55 | docelmo | So has anyone experienced muffled playback with asterisk? |
| 19:48.08 | mvanbaak | znoG_: there's this thing called phpagi |
| 19:48.16 | lesouvage | SIP implementation = SIP applications |
| 19:48.17 | *** join/#asterisk Shotygun (n=thorn@82.166.244.147) |
| 19:48.39 | znoG_ | mvanbaak/docelmo: yep, i'm looking at phpagi.sourceforge.net. I thought it was more for running PHP inside AGI applications but it looks like they designed it to be used standalone as well |
| 19:48.52 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 19:49.00 | docelmo | znoG_: yes we did.. :) Im one of the dev's |
| 19:49.28 | lwizardl | are there cards that are more supported for phone/fax use? |
| 19:49.37 | Shotygun | Hello. ringinuse prevents from a queue sending a call to a member who is already handling a queue call. However, it does not prevent from sending a call to a member which is busy with non-queue call. Is there a way to set a variable or something before the Dial() of the agent so app_queue will know the member is busy? |
| 19:50.12 | CoaxD | Shotygun: Welcome to phone-switch hell... That said, i have absolutely no useful feedback. |
| 19:50.16 | mvanbaak | <--- going to use it in his work |
| 19:50.29 | Shotygun | CoaxD Thank you =) |
| 19:50.30 | mvanbaak | docelmo: thank you for all the work :) |
| 19:50.43 | znoG_ | docelmo: nice work :) I noticed in the documentation, functions like mailboxcount() apparently return void but it says it returns the number of messages for that mailbox. Is that just an error in documentation? |
| 19:50.50 | docelmo | welcome.. Working on a new major release for 1.6 |
| 19:51.12 | znoG_ | docelmo: cool, i'm using 1.6 myself .. it should work OK, right? |
| 19:51.24 | docelmo | znoG_: I dont know off the top of my head.. I would have to research it. |
| 19:51.40 | docelmo | znoG_: yes but there might be some functions missing that 1.6 has that 2.14 doesnt support. |
| 19:51.46 | CoaxD | jesus, i'm still running 1.2.10 |
| 19:51.55 | docelmo | You would need to use the $agi->exec() function |
| 19:52.09 | znoG_ | docelmo: yep, that's cool. I'll either modify the class or, as you said, use $agi->exec() |
| 19:52.18 | docelmo | znoG_: for what you want to do might I suggess just using the manager interface to get what you need. |
| 19:52.35 | znoG_ | docelmo: manager interface? |
| 19:52.46 | docelmo | AMI |
| 19:52.47 | znoG_ | docelmo: not the AGI_AsteriskManager class? |
| 19:52.52 | docelmo | yes |
| 19:53.02 | znoG_ | docelmo: oh, right .. that's what I was planning to use .. the class |
| 19:53.14 | docelmo | Yes.. Use the PHPAGI AMI class |
| 19:53.27 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 19:53.29 | SamuraiDio | hi all |
| 19:53.52 | mintee | when i dial into my softphone, it comes up as ${CALLID(num)}@${ASTERISKIP} |
| 19:54.01 | Shotygun | CoaxD: In case you care, pausequeuemember() I think is the answer for it |
| 19:54.10 | mintee | i'm tried changing the domain settings in sip.conf |
| 19:54.27 | SamuraiDio | how do i connect an set my asterisk to receive connections without user auth? just by checking the ip address? |
| 19:54.29 | mintee | is there a way to have it come up as ${CALLID(num)}@DOMAINNAME |
| 19:54.29 | CoaxD | Shotygun: Hmmm... Interesting... So much has changed since i had a real need to configure a new asterisk system :( |
| 19:54.41 | Shotygun | CoaxD: Tell me about it.. |
| 19:54.45 | CoaxD | Shotygun: hell, my configs still use the old format! |
| 19:54.58 | Shotygun | likewise |
| 19:55.12 | znoG_ | docelmo: you speak spanish or you just threw some spanish in the doc? :) |
| 19:55.31 | CoaxD | Shotygun: Thing just runs and runs, though.... never fails unless the box fails... Or zaptel pukes, one of the two. Asterisk never fails. |
| 19:55.33 | docelmo | There is a team of 5 or so of us |
| 19:56.24 | Shotygun | CoaxD: I think they rather you'll refer to it as your "Dahdi" now.. |
| 19:56.41 | CoaxD | docelmo: When you can figure out the exact byte-payload that a FACILITY message to an avaya G3 contains in order to do an EECT transfer..... :) |
| 19:56.52 | Qwell | Shotygun: no, Zaptel and DAHDI are different things. |
| 19:56.55 | Qwell | sort of |
| 19:57.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:57.10 | Qwell | if somebody installed Zaptel, and they call it DAHDI, things could get very confused. |
| 19:57.31 | jjshoe | lol |
| 19:57.37 | jjshoe | who's your dahdi? |
| 19:57.47 | CoaxD | Qwell: hah. |
| 19:57.51 | russellb | be careful, the dahdi monitor is running |
| 19:57.54 | Shotygun | True, but I don't think that's far from reality |
| 19:58.02 | Shotygun | dahdi monitor? |
| 19:58.08 | Qwell | russellb: SHH |
| 19:58.09 | russellb | ztmonitor -> dahdi_monitor ... |
| 19:58.11 | russellb | OH |
| 19:58.13 | Shotygun | =D |
| 19:58.20 | CoaxD | installs mahmi |
| 19:58.21 | sp00kz | lol |
| 19:58.44 | sp00kz | bahbi monitor next? :/ |
| 19:58.56 | mvanbaak | or bahbi preventor ? |
| 19:58.58 | CoaxD | rolls eyes |
| 19:59.28 | russellb | runs dahdi_cfg on Qwell |
| 19:59.39 | mvanbaak | it will fail |
| 19:59.43 | mvanbaak | believe me |
| 19:59.47 | Shotygun | I am waiting for the first to come here & say "I'm having issues with my dahdi" |
| 19:59.58 | mvanbaak | Shotygun: yeah |
| 20:00.12 | Qwell | somebody has dahdi issues |
| 20:00.14 | russellb | trademarks are silly. |
| 20:00.21 | sp00kz | that name is silly |
| 20:00.22 | sp00kz | :| |
| 20:00.35 | Shotygun | Digium needs to hire a copyrighter |
| 20:00.42 | russellb | but not trademarked in the telephony space! |
| 20:00.50 | russellb | yet. |
| 20:01.01 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:01.16 | Shotygun | It's only trademarked among family consultants |
| 20:01.17 | mvanbaak | still, the name is a bit weird |
| 20:01.40 | russellb | mvanbaak: you need to get a dahdi_test |
| 20:01.54 | mvanbaak | russellb: let's hope not ! |
| 20:01.59 | russellb | see if your dahdi is accurate |
| 20:02.02 | russellb | timing, that is. |
| 20:02.06 | hsv-AL | nice, more 3am sirens expected tonight |
| 20:02.10 | hsv-AL | hate this crap |
| 20:02.16 | russellb | seriously? ugh |
| 20:02.37 | mvanbaak | so do we need a dahdi for conferencing ? |
| 20:02.41 | jjshoe | 3am sirens? |
| 20:02.44 | hsv-AL | Ts |
| 20:02.45 | Qwell | jjshoe: tornado |
| 20:02.45 | outtolunc | weird is fairly normal around here <G> |
| 20:03.04 | mvanbaak | gheh, this name is stupid. cant say a line with it in it without laughing |
| 20:03.06 | Qwell | I much prefer tornado sirens to a doom whistle... |
| 20:03.25 | russellb | mvanbaak: that's part of the point, heh |
| 20:03.28 | Shotygun | godahdi |
| 20:03.28 | Qwell | I used to live right near an active airforce base (< 5 miles). It went off...once. |
| 20:03.29 | jjshoe | Qwell ahh |
| 20:03.38 | hsv-AL | qwell, turn on whnt/waff at 2-3am, watch those casters "roll the sleeves" up |
| 20:03.39 | russellb | weather.com doesn't claim any bad weather for tonight ... |
| 20:03.44 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-66.static.insightbb.com) |
| 20:03.45 | hsv-AL | they practically get sexual thrills when stuff like this happens |
| 20:03.49 | hsv-AL | loves to make the public paranoid |
| 20:03.50 | Qwell | needless to say, people were scared :D |
| 20:03.50 | jjshoe | I've been through two tornado's, that's enough for me, now I'm trying out earthquakes |
| 20:03.58 | Shotygun | init.d/godahdi |
| 20:04.11 | mvanbaak | init.d/dahdi die |
| 20:04.15 | mvanbaak | init.d/dahdi resurrect |
| 20:04.30 | hsv-AL | I've never seen one with naked eye, but i do know a mile away from me |
| 20:04.30 | russellb | to find out who's your dahdi, run dahdi_scan |
| 20:04.37 | hsv-AL | like 20 people died a few years ago in a strip mall from one |
| 20:04.42 | *** join/#asterisk juice_d (n=juice_d@gozur.sunflowerbroadband.com) |
| 20:04.55 | znoG_ | dahdi is far more stable than mahmi |
| 20:05.07 | jjshoe | it's alabama, doesn't half the state live in tornado fuel? I mean trailer houses? |
| 20:05.07 | mvanbaak | and for the ones still living with their parents there will be a dahdi_ignore binary |
| 20:05.10 | hsv-AL | znog, are you a papi? waiting for your mami |
| 20:05.12 | Shotygun | mahmi needs to be the development tree |
| 20:05.12 | outtolunc | lilsis |
| 20:05.48 | outtolunc | ok ok that was bad |
| 20:05.49 | hsv-AL | time to get my $400 1985 buick regal, with $5,000 30" rims with reverse spinners, and chrome exhaust |
| 20:06.34 | sp00kz | hott |
| 20:06.51 | sp00kz | ghetto edition |
| 20:06.52 | sp00kz | ; |
| 20:06.54 | sp00kz | ;/ |
| 20:06.55 | hsv-AL | theres one that goes around town here, just like that with gothic fonts on the back window |
| 20:06.59 | hsv-AL | "Mami Boriqua" |
| 20:12.01 | docelmo | So has anyone in here experienced issues with muffled audio on asterisk audio files? |
| 20:13.26 | hsv-AL | spookz, you like? |
| 20:13.29 | hsv-AL | http://02civ.com/images/DONKeykong%2040%20inch%20rims.jpg |
| 20:13.36 | jjshoe | hsv-AL this car is at the body shop where my car is at currently http://i212.photobucket.com/albums/cc298/ocrnc/Donk.jpg |
| 20:13.56 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 20:14.18 | hsv-AL | those people that spend that kind of money on their cars, understand nothing about depreciation. |
| 20:14.30 | hsv-AL | Rims depecriate like 80% in 4-5 months :) |
| 20:14.30 | jjshoe | cars like that typically don't depreciate |
| 20:15.15 | outtolunc | love those mercedes rims <G> |
| 20:18.44 | docelmo | Can anyone in the USA dial this number: 1(866)502-8310 and tell me what they think the problem is with the audio? |
| 20:20.20 | *** join/#asterisk wichitalineman (n=svf@67.106.246.227.ptr.us.xo.net) |
| 20:21.50 | docelmo | ok see people dialing.. Any ideas? |
| 20:22.07 | docelmo | I have NEVER experienced this before with asterisk and cant figure out what the hell it is.. |
| 20:23.42 | rob0 | I dialed, but I have no clue, sorry. It sure is bad though. |
| 20:23.43 | jameswf-home | Digium folks... can we get an audio recording of either Allison or Mark saying Hi this is ________ and I pronounce Dahdi Dahdi |
| 20:24.06 | putnopvut | jameswf-home: lol |
| 20:24.07 | Qwell | jameswf-home: I will pass on the request. |
| 20:24.19 | docelmo | I dont get it.. |
| 20:24.22 | Qwell | (seriously too :p) |
| 20:24.33 | sp00kz | do it! |
| 20:24.41 | rob0 | "I prounounce doddy daydi!" |
| 20:24.43 | docelmo | so Qwell any ideas since yer the grand master @ Digium? |
| 20:24.59 | Qwell | what? |
| 20:25.06 | Qwell | oh |
| 20:25.06 | rob0 | "I now pronounce you Allison and Mark!" |
| 20:25.09 | jjshoe | Qwell is the grand master of my <3 |
| 20:25.17 | Qwell | <3 |
| 20:25.18 | docelmo | call this # : 8665028310 |
| 20:25.21 | Qwell | calling |
| 20:25.30 | Qwell | docelmo: play a different prompt |
| 20:25.39 | Qwell | like demo-congrats or something |
| 20:25.44 | docelmo | ok 1 sec.. |
| 20:26.16 | sp00kz | or tt-monkeysintro |
| 20:26.20 | docelmo | try now |
| 20:26.39 | Qwell | "password" ? |
| 20:26.50 | Qwell | something a little longer, and not monkeys :p |
| 20:27.02 | docelmo | ok I Will loop a couple through |
| 20:27.07 | outtolunc | muffled monkeys eh |
| 20:27.10 | Qwell | demo-congrats is perfect, IMO |
| 20:27.23 | sp00kz | you-sound-cute.wav |
| 20:27.24 | sp00kz | :( |
| 20:27.54 | sp00kz | first time I got the extra sound pack, i think i was laughing for a week |
| 20:28.45 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 20:28.47 | docelmo | ok demo-congrats & demo-instruct is loaded.. |
| 20:29.03 | Qwell | better |
| 20:29.13 | jameswf-home | wait found it http://www.dailymotion.com/video/x1f8yf_benny-benassi-whos-your-daddy-uncen_music |
| 20:29.22 | Qwell | docelmo: what version of asterisk? what codec(s) is it going through? |
| 20:29.49 | jameswf-home | OH CRAP NSFW :) |
| 20:30.27 | docelmo | g729 w/ gsm file and 1.4.19.2 |
| 20:30.32 | Qwell | hmm |
| 20:30.37 | docelmo | I did have ztdummy loaded but nixed it for testing |
| 20:30.39 | Qwell | jameswf-home: stay tuned. |
| 20:30.50 | docelmo | doesnt make a difference.. If I do ulaw/ulaw it doesnt matter either.. |
| 20:30.53 | docelmo | same results |
| 20:30.58 | docelmo | its kinda mind boggling |
| 20:31.06 | Qwell | with no transcoding? |
| 20:31.32 | docelmo | nope.. TDM(PRI) -> G729 -> Asterisk (playing a gsm file) |
| 20:31.34 | exothermc | is there someway to stop asterisk from asking for authentication after a client is registered? |
| 20:31.35 | Qwell | what kind of connection was that over, SIP? |
| 20:31.37 | docelmo | lemme switch to ulaw |
| 20:31.44 | Qwell | ahh, pri |
| 20:32.07 | exothermc | I turned insecure=invite,port on but it still asks for proxy auth on invites. |
| 20:32.54 | hsv-AL | heh, another 747 crashes into a sky scraper |
| 20:32.56 | hsv-AL | msnbc.com |
| 20:32.57 | docelmo | qwell ok playing ulaw now |
| 20:33.07 | docelmo | same thing |
| 20:33.23 | Qwell | no transcoding at all? still gsm prompts? |
| 20:33.41 | docelmo | no transcoding period.. ulaw prompts |
| 20:33.57 | Qwell | how are you connecting to the PRI? hardware in the Asterisk box? |
| 20:34.38 | Qwell | the audio is consistently crappy. |
| 20:35.17 | docelmo | no.. |
| 20:35.20 | jjshoe | hsv-AL you might want to refresh. |
| 20:35.22 | docelmo | gateway |
| 20:35.26 | Qwell | sip? |
| 20:35.29 | docelmo | yes |
| 20:35.34 | docelmo | audio sounds fine with on a live call |
| 20:35.47 | docelmo | or EVEN if you record a audio file then playback its fine.. |
| 20:35.58 | docelmo | Playing the canned audio files however sound like crap |
| 20:36.18 | *** join/#asterisk s0lid (n=s0lid@58.69.3.185) |
| 20:36.19 | Qwell | how do they sound with play/sox? |
| 20:36.34 | docelmo | you mean outside of asterisk? |
| 20:36.38 | Qwell | yeah |
| 20:36.57 | docelmo | havent tried.. but I did try the sounds from gsm, ulaw in 1.4 and 1.6 beta 9 |
| 20:37.06 | docelmo | but lemme try real quick I will tell ya |
| 20:38.18 | *** join/#asterisk Nasra (n=maxshipp@190.166.71.163) |
| 20:40.40 | docelmo | It sounds fine |
| 20:40.43 | docelmo | no issues |
| 20:40.59 | Qwell | and in the ulaw test, are you sure it was playing the .ulaw prompt? |
| 20:42.19 | docelmo | 99.9 being it was all ulaw asterisk should have used that one.. I will delete those gsm prompts |
| 20:43.51 | docelmo | interesting.. it was still picking up the file from .gsm.. |
| 20:44.02 | docelmo | I have ONLY ulaw in there and its telling me it cant find it now |
| 20:44.14 | Qwell | you sure you have the right dir? |
| 20:44.30 | docelmo | ./var/lib/asterisk/sounds yes |
| 20:44.34 | Qwell | check both /var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/en/ |
| 20:48.53 | *** join/#asterisk s0lid (n=s0lid@58.69.103.173) |
| 20:49.45 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:51.07 | docelmo | I cant find the sticking audio files in ulaw.. gsm yes.. So I changed them |
| 20:52.02 | docelmo | its working now.. it doesnt fucking like gsm for shit |
| 20:53.17 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:54.17 | docelmo | eh ohh well |
| 20:54.24 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 20:57.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:06.20 | *** join/#asterisk km- (n=pgrace@fmservices.ipv6.fierymoon.com) |
| 21:06.36 | km- | Is there any way to query the digium T1 cards to report framing errors? |
| 21:07.00 | km- | zttool will show the number of "bipolar violations" but it doesnt really say if there's any particular type of error. |
| 21:08.23 | tzanger | km-: which t1 card |
| 21:08.48 | tzanger | the te405/410/etc ones with the quadfalc do have the capablity in the driver... whether it is exposed through zaptel is another question entirely |
| 21:09.59 | km- | T220P PCIe 2 span T1 card |
| 21:10.09 | tzanger | not familliar with that one |
| 21:10.22 | tzanger | but check the driver, that's wehre I discovered the quadfalc's stuff |
| 21:10.26 | *** join/#asterisk Bananaskin (n=mike@user-5444d76a.lns1-c11.dsl.pol.co.uk) |
| 21:11.39 | km- | tzanger: know what it might be called in the driver to save me some grepping? |
| 21:11.54 | tzanger | look at the interrupt handler |
| 21:12.05 | tzanger | the HDLC controller stuff will also tell you things that zaptel doesn't know |
| 21:12.42 | km- | in this case we're trying to figure out if the card is getting timing slips |
| 21:13.27 | Bananaskin | [TK]D-Fender, u there ? |
| 21:14.48 | [TK]D-Fender | Bananaskin, yup |
| 21:16.07 | CCFL_Man2 | you guys see that silly majic jack infomercial? |
| 21:16.15 | *** join/#asterisk cjk_ (n=cjk@88.207.137.195) |
| 21:16.36 | *** join/#asterisk znoG (n=gs@host155.190-30-153.telecom.net.ar) |
| 21:16.48 | cjk_ | hi, how can i dial 100 sip and iax channs simultaniously? seems that i can not just concat them with & |
| 21:16.48 | Bananaskin | [TK]D-Fender, its ok, think I found what I am looking for.... will shout if it aint right (iaxmodem :)) |
| 21:16.58 | [TK]D-Fender | Bananaskin, yup, thats it |
| 21:18.19 | jjshoe | cjk I'm sure there's some limit to the number of them that can be passed in. |
| 21:18.29 | mvanbaak | wow |
| 21:18.43 | mvanbaak | I actually own 30x30 cm ground in Scotland |
| 21:18.54 | mvanbaak | 1 square foot |
| 21:18.54 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 21:19.16 | Bananaskin | would be worth more if it were cubed though :) |
| 21:19.28 | mvanbaak | http://images.vanbaak.info/displayimage.php?album=47&pos=0 |
| 21:20.05 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 21:20.39 | *** join/#asterisk marc7 (n=marc@w097-sub.wireless.uvic.ca) |
| 21:21.21 | anonymouz666 | [TK]D-Fender: why did you choose dahdi as a substitution name for zaptel? :P |
| 21:22.27 | exothermc | does polycom offer any video phones? |
| 21:22.42 | [TK]D-Fender | anonymouz666, My earlier statements are an unmistakable expression of my view of the new name. |
| 21:23.01 | anonymouz666 | smiles |
| 21:23.27 | exothermc | [TK]D-Fender: Found part of my issue for video |
| 21:24.15 | exothermc | [TK]D-Fender: Asterisk doesn't like to renegotiate the h263 apparently. I had to start the call with the video drawer open so it would negotiate h263 on call setup. |
| 21:25.30 | [TK]D-Fender | exothermc, that'd be the "check your client" part I spewed at you like 5-6 times :p |
| 21:25.50 | exothermc | [TK]D-Fender: eyebeam didn't want to respond to 407 proxy auth when the video drawer was open so I couldn't make a call out till I turned auth off in sip.conf then calling out with the video open works like a charm, but ya like I said can't seem to set it up once the first invite goes out if the invite doesn't contain the codec |
| 21:26.05 | exothermc | [TK]D-Fender: Well technically there is bug on both ends. |
| 21:26.36 | exothermc | [TK]D-Fender: Asterisk *should* still send h263 when that second invite comes down the pipe telling it about the new codec, but it does not. |
| 21:27.08 | exothermc | [TK]D-Fender: Yes I have something that works, but only if I hold my leg just right and lick my finger which isn't how I like to run. |
| 21:30.15 | exothermc | I have a support ticket into counterpath about the 407 issue with the drawer open, but I'm not sure what more information I need to gather to send to the asterisk dev team to get asterisk to renegotiate video mid session. |
| 21:30.41 | exothermc | hmm does asterisk do h264? |
| 21:30.48 | exothermc | or only 263. |
| 21:37.37 | *** join/#asterisk rgsteele||work (n=chatzill@74.94.57.81) |
| 21:39.29 | Wic3edOne | Does anyone here have experiance with * on a large scale ? |
| 21:39.44 | mvanbaak | exothermc: h263 AND h264 |
| 21:41.40 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:41.53 | grandpapadot | Hi all. Anyone know how to convert a ulaw to wav using sox? |
| 21:44.14 | exothermc | Wic3edOne: Most people that seen do things on a large scale don't use asterisk unless it is only for PBX functionality where you need a B2BUA then they tend to break it up into logical portions. |
| 21:44.38 | exothermc | **...that I've seen....** |
| 21:46.13 | *** part/#asterisk bkw_ (n=brian@adsl-71-153-169-69.dsl.tul2ok.sbcglobal.net) |
| 21:46.31 | Wic3edOne | i am working with a group of investors looking to statup a VOIP company using asterisk i have allready setup their business network and workstations and all that good stuff currently they are using another VOIP provider and just reselling accounts they want to move toward hosting aswell |
| 21:46.56 | CCFL_Man2 | Wic3edOne: investors |
| 21:47.01 | CCFL_Man2 | pashaw |
| 21:47.15 | Wic3edOne | i am really a Windows Network and Domain kinda guy |
| 21:47.32 | CCFL_Man2 | Wic3edOne: so is every other generic computer nerd |
| 21:47.32 | *** join/#asterisk marc7 (n=marc@p238-188.public.uvic.ca) |
| 21:47.33 | exothermc | Wic3edOne: Hosted PBX, residential gateway, enterprise trunking what type of end points are you targeting? |
| 21:47.59 | CCFL_Man2 | probably residential gateway |
| 21:48.08 | Wic3edOne | they are targeting small and medium businesses |
| 21:48.13 | CCFL_Man2 | ahh |
| 21:48.30 | CCFL_Man2 | Wic3edOne: a complete incoming and outgoing replacement? |
| 21:48.53 | exothermc | Wic3edOne: Then asterisk isn't a half bad choice, I would look into openvz and isolating the pbx environments in that. |
| 21:49.09 | Bananaskin | [TK]D-Fender, I assume I should be looking for something like - pbx FaxGetty[9812]: MODEM WWW.SOFT-SWITCH.ORG spandsp/ - in the output of var/log/messages |
| 21:50.09 | exothermc | Wic3edOne: Keeps your hardware overheads down, and keeps one pbx from borking the others. |
| 21:50.20 | [TK]D-Fender | Bananaskin, I never looked in the log files. jsut tested loading the modem "live", then under inittab after that "worked" |
| 21:51.18 | Bananaskin | Just last question, fax detection type (on my zaptel i/f's) - Zaptel, NVFAX or None ? |
| 21:51.32 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 21:52.21 | [TK]D-Fender | Bananaskin, I never personally bothered with detection. used "fax" for a client once though |
| 21:54.51 | Bananaskin | will try a couple of settings :) |
| 21:55.37 | CCFL_Man2 | yeah, ipkall works handily |
| 21:55.44 | CCFL_Man2 | nice |
| 21:57.00 | exothermc | CCFL_Man2: If you like them and you can generate a lot of traffic I can get you did which we'll pay $0.005/min for your traffic |
| 22:00.36 | CCFL_Man2 | exothermc: their did is free though |
| 22:00.53 | CCFL_Man2 | or don't they like lots of call traffic? |
| 22:01.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:02.22 | exothermc | CCFL_Man2: ya there is free, I'm saying we'll pay YOU for the traffic, no cost to you. |
| 22:06.06 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 22:07.10 | jjshoe | will I also get a potato slicer/dicer/peeler/flat tire fixer for free? |
| 22:07.16 | cjk_ | how would you gues dial 100 channels simultaniously (my sip users) its too much for the dial command |
| 22:08.15 | exothermc | jjshoe: Sorry it won't dice |
| 22:08.53 | mintee | Anyone know of a decent simple gsm sound recorder for windows? |
| 22:09.04 | mintee | hides his face for using the W word |
| 22:09.08 | exothermc | cjk: I have seen some weird things happens with dialing that many devices from within asterisk, but that has been a whlie. |
| 22:09.26 | exothermc | cjk: I would look into queues. |
| 22:09.37 | cjk_ | exothermc, ok thats what im doing right now |
| 22:09.43 | cjk_ | so i guess im on the right track |
| 22:10.20 | exothermc | cjk_: ya makes so much more sense anyway. Can't really think of a good scenario to ring 100 phones. |
| 22:10.40 | cjk_ | exothermc, i should now ring 25 and the limit is even there |
| 22:10.52 | cjk_ | imagine 25 users, all ahving a sipphone and an iax softphone |
| 22:10.58 | cjk_ | so 50 is reached quite fast |
| 22:11.30 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 22:11.31 | *** mode/#asterisk [+o twisted] by ChanServ |
| 22:12.50 | grandpapadot | Is there another way through queues.conf to specify a queue timeout vs using the Queue() cmd? |
| 22:13.15 | hardwire | you have to dial a queue right? |
| 22:13.37 | hardwire | maybe you could dial(Local/queextension@context,timeoutstuff) |
| 22:17.51 | *** join/#asterisk rcy` (n=rcy@S010600207810532a.vc.shawcable.net) |
| 22:27.32 | *** join/#asterisk marc7 (n=marc@p238-188.public.uvic.ca) |
| 22:30.59 | *** join/#asterisk klimonso (n=eddy@dxb-b12891.alshamil.net.ae) |
| 22:31.16 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:32.05 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:35.32 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 22:37.08 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 22:39.34 | mintee | is there anywhere the asterisk sounds are typed out? |
| 22:39.46 | mintee | specifically the vm-* sounds? |
| 22:40.16 | Qwell | core-sounds-en.txt |
| 22:40.29 | mintee | awesome, thanks Qwell. |
| 22:40.34 | mintee | You rock again! |
| 22:46.14 | deeperror | is ztdummy only available from svn? |
| 22:49.07 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 22:49.13 | *** join/#asterisk [hC] (n=hardcore@209.82.105.158) |
| 22:49.45 | _ShrikE | deeperror: no |
| 22:50.04 | deeperror | i see the source i'm following something on voip-info seems a bit outdated |
| 22:50.46 | [hC] | Anyone have any idea why a polycom ip430 would wait almost 15 minutes at "Updating BootROM" before beginning to fill in the progress bar? |
| 22:51.57 | *** join/#asterisk netstv (n=stv@phoenix.u4eatechinc.com) |
| 22:53.21 | netstv | anyone with expertise with mips cross compile? |
| 22:54.44 | netstv | anyone out there? |
| 22:54.55 | jaytee | netstv, hi |
| 22:55.28 | deeperror | n |
| 22:55.30 | jaytee | netstv, I've never run linux on mips hardware, just x86 |
| 22:55.44 | *** join/#asterisk timothy_jones (n=timothy_@207.90.4.158) |
| 22:57.37 | *** part/#asterisk twitchnln (n=twitch@c-76-105-88-129.hsd1.ga.comcast.net) |
| 22:58.04 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 22:58.32 | timothy_jones | hi there, came across highpriority=no setting in asterisk.conf. It says it will run * with realtime priority. Are there any side effects of setting it to "yes"? |
| 23:02.29 | timothy_jones | thinks: first time I am using epic as my irc client, not sure if the above actually got posted... |
| 23:03.44 | timothy_jones | ~asterisk |
| 23:03.44 | jbot | somebody said asterisk was the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 23:03.53 | tzafrir | timothy_jones, 'highpriority=yes' should probably have that effect, not 'no' |
| 23:03.58 | TrentCreek | ~freepbx |
| 23:03.58 | 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 |
| 23:06.06 | timothy_jones | hmmm, not sure why it would be set to no by default |
| 23:06.50 | Qwell | timothy_jones: in many cases, you might not want Asterisk running with higher priority |
| 23:08.48 | *** join/#asterisk adr3nalin3_ (n=afink@ip68-13-89-102.om.om.cox.net) |
| 23:09.00 | MaartenB | how can I debug dropped calls when doing an announced transfer on SPA942's? |
| 23:09.07 | timothy_jones | I guess if ur running it in realtime mode with database on the same pc, it could create side effects... otherwise I can't see any reason why wit would be bad, right? |
| 23:09.27 | *** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com) |
| 23:10.12 | *** part/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 23:12.15 | timothy_jones | I've got 8 * servers running 1.2 with sip trunks. All are stacking up dead channels. I upgraded one to 1.4 and it's better, but still getting them |
| 23:12.37 | shmaltz | timothy_jones, what do you mean by dead channels? |
| 23:14.10 | timothy_jones | eg, show channels shows 4 channels (2 people speaking on the phone) whilst sip show channels shows way more than 4 and they are all marked (d) |
| 23:15.17 | *** join/#asterisk sarthor (n=sarthor@88.81.15.223) |
| 23:16.41 | timothy_jones | shmaltz, u know what mean? |
| 23:17.08 | shmaltz | timothy_jones, no, I'm running 1.2.x all over, never seen this |
| 23:17.13 | shmaltz | checking some more machines |
| 23:17.38 | sarthor | Is it possible?? want to configure a voip server having pstn pci card in it, put that server in my counter, A, i am in counter B, and i want to call to my home from internet, and also want to call some of my frineds mobiles and fix fones, Will pstn card will help in this.. Is it possible????? |
| 23:17.38 | [hC] | the loading times on this IP430 on sip firmware 2.1.1 are just -insane-. I've been sitting at "running sip.ld" for roughly 15 minutes now. |
| 23:19.06 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20 (2008/05/20), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 23:19.15 | shmaltz | sarthor, read read read, from your questions I understand that you need to read a lot before you can even ask another question, no offense I was the same a while back |
| 23:19.17 | mcab | [hC]: are you running a bootserver? |
| 23:20.05 | sarthor | shmaltz, your advice is worthy, i did read, but ia m not much educated,, but i will try again.. Thank you.... |
| 23:20.32 | shmaltz | sarthor, you can start here: |
| 23:20.34 | maqr | anyone had decent luck with callcentric? |
| 23:20.35 | shmaltz | ~thebook |
| 23:20.35 | 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 --- HTML at http://tfot.leifmadsen.com |
| 23:21.09 | [hC] | mcab: what do you mean, a bootserver? I'm running a dhcp server which hands out option 66 and option 150 to find its FTP configuration server yes. |
| 23:21.38 | mcab | [hC]: ok, and the 430 can connect to the config server? |
| 23:21.51 | timothy_jones | maqr, we use them as one of our providers. they are not bad, they don't have phone number which sucks |
| 23:21.53 | [hC] | mcab: yes, it connects, it just takes -forever- to do it. |
| 23:21.58 | mcab | [hC]: IME long delays like that are the phone timing out trying ot get it's config |
| 23:22.09 | *** join/#asterisk golumn (n=acxty@201.220.132.138) |
| 23:22.24 | maqr | timothy_jones: hmm, ok, do you use their fax thing at all? |
| 23:22.54 | golumn | Hi, I have some recording in 44100Hz, which program can I use to use them on asterisk. What I did was sox -r 8000 but it distort the sound |
| 23:23.03 | timothy_jones | maqr, no I not use fax only phone |
| 23:23.11 | maqr | timothy_jones: oh, ok, ty :) |
| 23:23.52 | mcab | [hC]: shot in the dark, phone plugged into a bad port that's losing lots of packets? |
| 23:24.12 | *** part/#asterisk timothy_jones (n=timothy_@207.90.4.158) |
| 23:24.40 | [hC] | mcab: nope, they all do it (over 250 phones) - the dhcp server is simply configured to hand out option 66, pointing to my config server. now one thing im thinking is it might be trying tftp first and timing out |
| 23:25.40 | *** join/#asterisk joe_schmoe (n=timothy_@207.90.4.158) |
| 23:25.47 | joe_schmoe | ~callweaver |
| 23:25.48 | jbot | hmm... callweaver is something that started off as a fork of Asterisk (b the name of openpbx), but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in Asterisk. see http://callweaver.org/ for more info, or join #callweaver |
| 23:26.09 | *** join/#asterisk RoyK (n=roy@ip-199-56-149-91.dialup.ice.no) |
| 23:26.59 | *** part/#asterisk joe_schmoe (n=timothy_@207.90.4.158) |
| 23:27.13 | russellb | that is so far from accurate ... |
| 23:27.30 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:27.35 | *** join/#asterisk timothy_jones (n=timothy_@207.90.4.158) |
| 23:27.45 | Qwell | russellb: fix it |
| 23:29.55 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-184-133.dsl.stlsmo.sbcglobal.net) |
| 23:30.41 | jameswf-home | oh snap |
| 23:31.00 | seanbright | i think he joined the channel just to type ~callweaver |
| 23:31.07 | seanbright | people are weird. |
| 23:31.11 | russellb | heh, BURN |
| 23:31.26 | jameswf-home | I added ~dahdi and ~dahdijoke |
| 23:31.36 | seanbright | ~seanbright |
| 23:31.37 | jbot | somebody said seanbright was a girl with standards |
| 23:31.42 | jameswf-home | i need a hobby |
| 23:31.42 | seanbright | yay |
| 23:32.04 | LemensTS | I need a prepaid/postpaid solution with CDR, opensource for free. I will be doing things such as allowing them to replinish funds over the phone. What would be a good base application to start from? Thier are so many i would like someones opinion who has done this before...would save me hours |
| 23:32.29 | seanbright | wow... orioles are whipping the yankees |
| 23:32.32 | russellb | I need to make money without investing any. |
| 23:32.37 | russellb | Somehow that seems impossible |
| 23:32.39 | seanbright | 6-0 top of the *1st* |
| 23:32.42 | jameswf-home | ooh i want some of that too |
| 23:33.01 | Qwell | LemensTS: Asterisk seems like a good starting point. |
| 23:33.06 | seanbright | russellb: its called being employed, no? |
| 23:33.08 | russellb | Qwell: I agree. |
| 23:33.09 | murdock_ut | ~dahdijoke |
| 23:33.10 | jbot | i guess dahdijoke is http://files.jameswf.info/dahdihorse.jpg |
| 23:33.18 | Qwell | I'm afraid |
| 23:33.22 | russellb | seanbright: you're still investing resources at that point. |
| 23:33.25 | jameswf-home | heh |
| 23:33.26 | Qwell | nice |
| 23:33.36 | seanbright | russellb: right, but not money, per se. |
| 23:33.38 | Qwell | jbot: jameswf-home++ |
| 23:33.38 | murdock_ut | ha ha |
| 23:33.39 | *** join/#asterisk rsc-232 (n=mrdigita@65-78-101-58.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 23:33.49 | MaartenB | anyone having experienced in figuring out why calls are dropped on transfer? |
| 23:34.11 | LemensTS | Qwell: yes of course it is on asterisk. I mean thier are alot of addons that do most of this: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications |
| 23:34.50 | LemensTS | I would hate to have to install each one to check it out, it would be nice if someone has used some of them and could tell me if one is good, or if one sucks...etc |
| 23:35.19 | *** join/#asterisk blinky42 (n=blinky42@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:36.00 | Qwell | ~dahdi |
| 23:36.01 | jbot | [dahdi] Digium/Asterisk Hardware Device Interface. The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi |
| 23:36.20 | LemensTS | zaptel is dahdi now? huh |
| 23:36.39 | jameswf-home | LemensTS: yeah that is so last week dont you watch commits |
| 23:36.49 | jameswf-home | :) |
| 23:38.51 | LemensTS | so is zapata.conf gonna be dahdda.conf ? |
| 23:38.52 | LemensTS | heh |
| 23:39.09 | jameswf-home | no zaptel.conf |
| 23:39.15 | jameswf-home | if anything |
| 23:40.23 | timothy_jones | how do u pronounce it, [daddy] or [duh-di]? :) |
| 23:40.44 | russellb | dah-dee |
| 23:41.02 | russellb | jbot: dahdi is also pronounced "dah-dee" |
| 23:41.03 | jbot | russellb: okay |
| 23:41.14 | seanbright | imagine a german woman calling you 'daddy' |
| 23:41.16 | seanbright | heh |
| 23:41.31 | jameswf-home | <PROTECTED> |
| 23:41.36 | russellb | hrm, should have also indicated that it was a short A |
| 23:41.45 | russellb | jbot: forget dahdi |
| 23:41.45 | jbot | russellb: i forgot dahdi |
| 23:42.05 | russellb | jbot: dahdi is Digium/Asterisk Hardware Device Interface. The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi, and is pronounced "dah-dee" with a short A |
| 23:42.05 | jbot | russellb: okay |
| 23:42.08 | jameswf-home | this is why i suggested the recording earlier |
| 23:42.31 | Qwell | jameswf-home: I did ask Mark. |
| 23:42.53 | jameswf-home | something is lost in the phonetics |
| 23:43.10 | Qwell | he said he'd record something, but no idea when |
| 23:43.15 | Qwell | he's already forgotten, I'm sure :p |
| 23:43.21 | russellb | shrugs |
| 23:43.31 | seanbright | allison is in vegas anyway |
| 23:43.50 | russellb | it's as simple as a 1 line asterisk extension :) |
| 23:43.50 | jameswf-home | I got the new software on my blackberry so I can do videos on the spot :) |
| 23:44.41 | seanbright | i would have just called it 'letpaz' or 'telzap' |
| 23:44.50 | seanbright | i should be in marketing |
| 23:45.46 | jameswf-home | I would call it chiclets makes me think of mexico and the small children with the penny gum |
| 23:46.19 | seanbright | ohhh, or pr0n |
| 23:46.31 | seanbright | 'digium announces the release of pr0n 2.0.0' |
| 23:46.36 | seanbright | i'd download |
| 23:47.42 | russellb | jbot: dahdi is also pronounced like http://www.russellbryant.net/dahdi.wav |
| 23:47.42 | jbot | okay, russellb |
| 23:48.32 | *** join/#asterisk EvilRick (n=EvilRick@41.207.226.226) |
| 23:48.46 | timothy_jones | or as us germans would say, dutty |
| 23:48.54 | seanbright | dotty |
| 23:49.02 | EvilRick | hey guys, whats the best way to forward a specific did from a ZAP channel to another server? |
| 23:49.04 | putnopvut | russellb: you should have had a link to rickroll. |
| 23:49.21 | EvilRick | I am using trixbox and would like to do it via the web interface |
| 23:49.27 | seanbright | ~trixbox |
| 23:49.28 | 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 |
| 23:49.39 | seanbright | EvilRick: ^^^ |
| 23:50.11 | russellb | putnopvut: touche. |
| 23:50.15 | *** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:50.27 | *** join/#asterisk RoyK (n=roy@ip-199-56-149-91.dialup.ice.no) |
| 23:51.13 | EvilRick | ok, I guess. ok forget about txbx. lets say I wanted to do that on vanilla asterisk, how would I go about it |
| 23:51.17 | EvilRick | IAX trunk? |
| 23:51.41 | timothy_jones | iax is the way |
| 23:52.39 | timothy_jones | Inter Trixbox[silent A] eXchange ;) |
| 23:52.58 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 23:53.21 | jameswf-home | ooh russellb you should do 900 calls :) |
| 23:53.35 | russellb | O.O |
| 23:55.28 | hardwire | I'd only pay for more than one word. |
| 23:56.43 | seanbright | would like to reiterate that his o's are currently destroying the yankees |
| 23:57.51 | putnopvut | Go Orioles! |
| 23:58.30 | seanbright | woo! |
| 23:58.50 | seanbright | is hoping for a .500 season. that's all i want. |
| 23:59.03 | drmessano | Go yankees |
| 23:59.06 | drmessano | Wait |
| 23:59.07 | jjshoe | go aerioles! |
| 23:59.12 | drmessano | Screw the yankees |
| 23:59.17 | drmessano | They let Mattingly go |
| 23:59.22 | drmessano | YAY DODGERS |
| 23:59.39 | jjshoe | EvilRick which version of trixbox? |
